mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
2660 Commits (9c7c1897e55b479dfbed8cdc40acaadee32609fe)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
557125664d |
This patch adds support for controlling a playback operation from the
Asterisk REST interface.
This adds the /playback/{playbackId}/control resource, which may be
POSTed to to pause, unpause, reverse, forward or restart the media
playback.
Attempts to control a playback that is not currently playing will
either return a 404 Not Found (because the playback object no longer
exists) or a 409 Conflict (because the playback object is still in the
queue to be played).
This patch also adds skipms and offsetms parameters to the
/channels/{channelId}/play resource.
(closes issue ASTERISK-21587)
Review: https://reviewboard.asterisk.org/r/2559
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
|
10ba6bf8a8 |
This patch implements the REST API's for POST /channels/{channelId}/play
and GET /playback/{playbackId}.
This allows an external application to initiate playback of a sound on a
channel while the channel is in the Stasis application.
/play commands are issued asynchronously, and return immediately with
the URL of the associated /playback resource. Playback commands queue up,
playing in succession. The /playback resource shows the state of a
playback operation as enqueued, playing or complete. (Although the
operation will only be in the 'complete' state for a very short time,
since it is almost immediately freed up).
(closes issue ASTERISK-21283)
(closes issue ASTERISK-21586)
Review: https://reviewboard.asterisk.org/r/2531/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389587 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 |
|
|
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 |
13 years ago |
|
|
4e38a4eb64 |
Move origination to use the dialing API and send Stasis messages on dial begin and end.
(closes issue ASTERISK-21549) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2512/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
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 |
13 years ago |
|
|
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 |
13 years ago |
|
|
b90bba7a30 |
Stasis: Update security events to use Stasis
Also moves ACL messages to the security topic and gets rid of the ACL topic (closes issue ASTERISK-21103) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2496/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388975 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
1ead1853f2 |
Use srtp_shutdown when available
This allows the SRTP library to be shut down properly when the functionality is offered by libsrtp. Review: https://reviewboard.asterisk.org/r/2538/ (closes issue ASTERISK-21719) ........ Merged revisions 388768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 388769 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
9648e258c7 |
Refactored the rest of the message types to use the STASIS_MESSAGE_TYPE_*
macros. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388751 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
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 |
13 years ago |
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d1d1425327 |
Make ao2 global objects not always use the debug version of the ao2_ref() calls.
The debug versions of ao2_ref() should only be used if REF_DEBUG is
enabled so nothing is written to /tmp/refs unexpectedly.
(closes issue ASTERISK-21785)
Reported by: abelbeck
Patches:
jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: abelbeck
........
Merged revisions 388700 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
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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 |
13 years ago |
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db925c3f06 |
Avoided __ast names for the private variables created by the
STASIS_MESSAGE_TYPE_*() macros. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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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 |
13 years ago |
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b007c744a4 |
Add development flag to disable the inline API.
A GCC bug[1] can, in some cases, pop up an unsuppressible pedwarn when using a static inline standard library function from a non-static inline function. This normally doesn't show up, but can occur if you're running an upgrade version of GCC (such as GCC 4.8 on OS X, which normally runs GCC 4.2). [1]: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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0eb4cf8c19 |
Remove required type field from channel blobs
When we first introduced the channel blob types, the JSON blobs were self identifying by a required "type" field in the JSON object itself. This, as it turns out, was a bad idea. When we introduced the message router, it was useless for routing based on the JSON type. And messages had two type fields to check: the stasis_message_type() of the message itself, plus the type field in the JSON blob (but only if it was a blob message). This patch corrects that mistake by removing the required type field from JSON blobs, and introducing first class stasis_message_type objects for the actual message type. Since we now will have a proliferation of message types, I introduced a few macros to help reduce the amount of boilerplate necessary to set them up. Review: https://reviewboard.asterisk.org/r/2509 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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e06e519a90 |
Initial support for endpoints.
An endpoint is an external device/system that may offer/accept channels to/from Asterisk. While this is a very useful concept for end users, it is surprisingly not a core concept within Asterisk itself. This patch defines ast_endpoint as a separate object, which channel drivers may use to expose their concept of an endpoint. As the channel driver creates channels, it can use ast_endpoint_add_channel() to associate channels to the endpoint. This updated the endpoint appropriately, and forwards all of the channel's events to the endpoint's topic. In order to avoid excessive locking on the endpoint object itself, the mutable state is not accessible via getters. Instead, you can create a snapshot using ast_endpoint_snapshot_create() to get a consistent snapshot of the internal state. This patch also includes a set of topics and messages associated with endpoints, and implementations of the endpoint-related RESTful API. chan_sip was updated to create endpoints with SIP peers, but the state of the endpoints is not updated with the state of the peer. Along for the ride in this patch is a Stasis test API. This is a stasis_message_sink object, which can be subscribed to a Stasis topic. It has functions for blocking while waiting for conditions in the message sink to be fulfilled. (closes issue ASTERISK-21421) Review: https://reviewboard.asterisk.org/r/2492/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387932 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a61060cf41 |
Fixed up \example marker in lock.h Doxygen comment.
The \example tags marks an entire file as an example, not a code snippet. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387825 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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737a45f2f7 |
Better explained the depths of reference stealing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387803 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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570d0c3139 |
Fix build breakage, from LOW_MEMORY fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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1beb86ddf5 |
Update ao2_destructor_fn doxygen.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387741 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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40074542bf |
Add support for observers and JSON objectset creation to sorcery.
This change adds the ability for modules to add themselves as observers to sorcery object types. Observers can be notified when objects are created, updated, or deleted as well as when the object type is loaded or reloaded. Observer notifications are done using a thread pool in a serialized fashion so the caller of the sorcery API calls is minimally impacted. This also adds the ability to create JSON changesets of a sorcery object. Tests are also present to confirm all of the above functionality. Review: https://reviewboard.asterisk.org/r/2477/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387662 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6e2fe0c9ab |
Clean up documentation; prevent ref leak on exit
This patch: * Cleans up some doxygen * Prevents leaking the system level Stasis topics and messages on exit (users of valgrind will be happier) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387633 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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1eac5a7988 |
Stasis: Convert network change events into network change stasis messages
(issue ASTERISK-21103) Review: https://reviewboard.asterisk.org/r/2490/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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02961601cd |
Putting all event defs and names back for now due to res_corosync dependency
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387420 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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8e257fe819 |
Stasis Core: Refactor ACL Change events to go out over the stasis core msg bus
(issue ASTERISK-21103) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2481/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387037 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c03b95a3d7 |
Use the proper lower bound when doing saturation arithmetic.
16 bit signed integers have a range of [-32768, 32768). The existing code was using the interval (-32768, 32768) instead. This patch fixes that. Review: https://reviewboard.asterisk.org/r/2479/ ........ Merged revisions 386929 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 386930 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386931 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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fc47a3f6e5 |
Change pointer to existing wiki page instead of non-existing page
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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02be50b1ac |
Add support for a realtime sorcery module.
This change does the following: 1. Adds the sorcery realtime module 2. Adds unit tests for the sorcery realtime module 3. Changes the realtime core to use an ast_variable list instead of variadic arguments 4. Changes all realtime drivers to accept an ast_variable list Review: https://reviewboard.asterisk.org/r/2424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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946eb5ede0 |
Example of how to use the Stasis message bus
In order to get people familiar with the Stasis message bus, it would be useful to have something of a tutorial. Since I'm not clever enough to think of some cool integration we could do with Twitter, I settled for something that might actually be useful. This patch adds a res_statsd.so module, which implements a basic statsd[1] client. Statsd is a very simple statistics gathering server, which can publish its results to a backend graphing engine, like Graphite[2]. There are several different Statsd server implementations[3], so you can pick what works best for your environment. The actual example of how to use the Stasis message bus is in res_chan_stats.so. This module demonstrates how to use subscriptions and the message router by monitoring messages and posting channels stats to the statsd server. A wiki page walking through res_chan_stats.so is forthcoming. [1]: https://github.com/etsy/statsd/ [2]: http://graphite.readthedocs.org/en/latest/ [3]: http://joemiller.me/2011/09/21/list-of-statsd-server-implementations/ Review: https://reviewboard.asterisk.org/r/2460/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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74f2318051 |
Merge the pimp_my_sip branch into trunk.
The pimp_my_sip branch is being merged at this point because it offers basic functionality, and from an API standpoint, things are complete. SIP work is *not* feature-complete; however, with the completion of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have been created, and thus it is possible for developers to attempt to create new SIP work. API documentation can be found in the doxygen in the code, but usability documentation is still lacking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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e05870ccae |
Doxygen - Markup Guidelines
Expand on a commit by OEJ to use the Coding-Guidelines (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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1c21b8575b |
This patch adds a RESTful HTTP interface to Asterisk.
The API itself is documented using Swagger, a lightweight mechanism for documenting RESTful API's using JSON. This allows us to use swagger-ui to provide executable documentation for the API, generate client bindings in different languages, and generate a lot of the boilerplate code for implementing the RESTful bindings. The API docs live in the rest-api/ directory. The RESTful bindings are generated from the Swagger API docs using a set of Mustache templates. The code generator is written in Python, and uses Pystache. Pystache has no dependencies, and be installed easily using pip. Code generation code lives in rest-api-templates/. The generated code reduces a lot of boilerplate when it comes to handling HTTP requests. It also helps us have greater consistency in the REST API. (closes issue ASTERISK-20891) Review: https://reviewboard.asterisk.org/r/2376/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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1871017cc6 |
Fix mistake in Doxygen.
Doxygen is only *ONE* comment that applies to the NEXT piece of code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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9c696e665f |
Allow WebSocket connections on more URL's
This patch adds the concept of ast_websocket_server to res_http_websocket, allowing WebSocket connections on URL's more more than /ws. The existing funcitons for managing the WebSocket subprotocols on /ws still work, so this patch should be completely backward compatible. (closes issue ASTERISK-21279) Review: https://reviewboard.asterisk.org/r/2453/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386020 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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71a01725b8 |
Move presence state distribution to Stasis-core
Convert presence state events to Stasis-core messages and remove redundant serializers where possible. Review: https://reviewboard.asterisk.org/r/2410/ (closes issue ASTERISK-21102) Patch-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385862 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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191cf99ae1 |
Move device state distribution to Stasis-core
In the move from Asterisk's event system to Stasis, this makes distributed device state aggregation always-on, removes unnecessary task processors where possible, and collapses aggregate and non-aggregate states into a single cache for ease of retrieval. This also removes an intermediary step in device state aggregation. Review: https://reviewboard.asterisk.org/r/2389/ (closes issue ASTERISK-21101) Patch-by: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c1ae5dc49b |
Fixed a typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385835 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c599aca553 |
Moved core logic from app_stasis to res_stasis
After some discussion on asterisk-dev, it was decided that the bulk of
the logic in app_stasis actually belongs in a resource module instead
of the application module.
This patch does that, leaves the app specific stuff in app_stasis, and
fixes up everything else to be consistent with that change.
* Renamed test_app_stasis to test_res_stasis
* Renamed app_stasis.h to stasis_app.h
* This is still stasis application support, even though it's no
longer in an app_ module. The name should never have been tied to
the type of module, anyways.
* Now that json isn't a resource module anymore, moved the
ast_channel_snapshot_to_json function to main/stasis_channels.c,
where it makes more sense.
Review: https://reviewboard.asterisk.org/r/2430/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
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2450722f52 |
DTMF events are now published on a channel's stasis_topic. AMI was
refactored to use these events rather than producing the events directly in channel.c. Finally, the code was added to app_stasis to produce DTMF events on the WebSocket. The AMI events are completely backward compatible, including sending events on transmitted DTMF, and sending DTMF start events. The Stasis-HTTP events are somewhat simplified. Since DTMF start and DTMF send events are generally less useful, Stasis-HTTP will only send events on received DTMF end. (closes issue ASTERISK-21282) (closes issue ASTERISK-21359) Review: https://reviewboard.asterisk.org/r/2439 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385734 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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ab6c0e74f1 |
Fix documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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7f885dc31d |
Expose channel snapshot manager blob generation
These functions are already used in one branch (jrose's parking branch) and will soon be used in other branches as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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b8d4e573f1 |
Add multi-channel Stasis messages; refactor Dial AMI events to Stasis
This patch does the following: * A new Stasis payload has been defined for multi-channel messages. This payload can store multiple ast_channel_snapshot objects along with a single JSON blob. The payload object itself is opaque; the snapshots are stored in a container keyed by roles. APIs have been provided to query for and retrieve the snapshots from the payload object. * The Dial AMI events have been refactored onto Stasis. This includes dial messages in app_dial, as well as the core dialing framework. The AMI events have been modified to send out a DialBegin/DialEnd events, as opposed to the subevent type that was previously used. * Stasis messages, types, and other objects related to channels have been placed in their own file, stasis_channels. Unit tests for some of these objects/messages have also been written. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a2a53cc306 |
Stasis application WebSocket support
This is the API that binds the Stasis dialplan application to external Stasis applications. It also adds the beginnings of WebSocket application support. This module registers a dialplan function named Stasis, which is used to put a channel into the named Stasis app. As a channel enters and leaves the Stasis diaplan application, the Stasis app receives a 'stasis-start' and 'stasis-end' events. Stasis apps register themselves using the stasis_app_register and stasis_app_unregister functions. Messages are sent to an application using stasis_app_send. Finally, Stasis apps control channels through the use of the stasis_app_control object, and the family of stasis_app_control_* functions. Other changes along for the ride are: * An ast_frame_dtor function that's RAII_VAR safe * Some common JSON encoders for name/number, timeval, and context/extension/priority Review: https://reviewboard.asterisk.org/r/2361/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384879 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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bcc0aca23d |
Make things work again
Sorry folks. ',' are still greater than '|'. Thanks for playing along :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384514 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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8c5367226b |
Make appropriate items parse using '|' instead of ','
This patch fixes a bug introduced in r76703, wherein Asterisk could only parse arguments in the so-called 'recommended' way, e.g., NoOp(foo,bar). The proper syntax of NoOp,foo|bar is now parsed correctly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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e8015cc460 |
Convert TestEvent AMI events over to Stasis Core
This patch migrates the TestEvent AMI events to first be dispatched over the Stasis-Core message bus. This helps to preserve the ordering of the events with other events in the AMI system, such as the various channel related events. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384389 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a1c94fece8 |
Add uuid wrapper API call ast_uuid_generate_str().
* Updated test_uuid.c to test the new API call. * Made system use the new API call to eliminate "10's of lines" where used. * Fixed untested ast_strdup() return in stasis_subscribe() by eliminating the need for it. struct stasis_subscription now contains the uniqueid[] string. * Fixed some issues in exchangecal_write_event(): Create uid with enough space for a UUID string to avoid a realloc. Fix off by one error if the calendar event provided a UUID string. There is no need to check for NULL before calling ast_free(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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71206544a7 |
Break the world. Stasis message type accessors should now all be named correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |