mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
2687 Commits (3421960162a4857eb365dd87ccf4cd6202643cd4)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
c43f380d03 |
Add backtrace generation to MALLOC_DEBUG memory corruption reports
This patch allows astmm to access the backtrace generation code in Asterisk. When memory is allocated, a backtrace is created and stored with the memory region that tracks the allocation. If a memory corruption is detected, the backtrace is printed to the astmm log. The backtrace will make use of the BETTER_BACKTRACES build option if available. As a result, this patch moves the backtrace generation code into its own file and uses the non-wrapped versions of the C library memory allocation routines. This allows the memory allocation code to safely use the backtrace generation routines without infinitely recursing. Review: https://reviewboard.asterisk.org/r/2567 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391012 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
2fe6b6a533 |
Add more support for native bridging.
* Added a start technology callback that technologies can use to start bridging operations. It is expected that native bridges will find this useful. * Factored out bridge_channel_complete_join(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
c88b7945f6 |
Fix a crash when a bridge switches from the softmix bridge technology to another.
A three party bridge uses the softmix bridging technology. This technology has a dedicated thread used to perform the analog mixing. When one of these parties leaves the bridge, the bridge technology is changed from the softmix technology to a two-party mixing technology. Changing technologies is done by removing channels from the old technology and adding them to the new technology. Since the remaining channels do not leave the bridge, the softmix mixing thread could continue to process all channels in the bridge. If the bridge code is not able to start destruction of the softmix technology before the softmix mixing thread wakes up, a crash happens. * Added a stop technology callback that technologies can use to request any helper threads to stop in preparation for being destroyed. (closes issue AST-1156) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390975 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
661f6d499e |
Update some doxygen comments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
8954661207 |
res_parking: Automatically generate extensions, hints, etc.
(closes issue ASTERISK-21645) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2545/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390849 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 |
|
|
759a7e4a30 |
Rework stasis cache clear events
Stasis cache clear message payloads now consist of a stasis_message representative of the message to be cleared from the cache. This allows multiple parallel caches to coexist and be cleared properly by the same cache clear message even when keyed on different fields. This change fixes a bug where multiple cache clears could be posted for channels. The cache clear is now produced in the destructor instead of ast_hangup. Additionally, dummy channels are no longer capable of producing channel snapshots. Review: https://reviewboard.asterisk.org/r/2596 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
bad8caa8c6 |
Reimplement bridging and DTMF features related channel variables in the bridging core.
* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel driver specific. If the channel variable is set on the transferrer channel, the sound will be played to the target of an attended transfer. * The channel variable BRIDGEPEER becomes a comma separated list of peers in a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers listed. Any more peers in the bridge will not be included in the list. BRIDGEPEER is not valid in holding bridges like parking since those channels do not talk to each other even though they are in a bridge. * The channel variable BRIDGEPVTCALLID is only valid for two party bridges and will contain a value if the BRIDGEPEER's channel driver supports it. * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that activated the dynamic feature. * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set only on the channel executing the dynamic feature. Executing a dynamic feature on the bridge peer in a multi-party bridge will execute it on all peers of the activating channel. (closes issue ASTERISK-21555) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2582/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 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 |
|
|
9f54568010 |
Convert message_router routes to ao2. Add support for removal.
Review: https://reviewboard.asterisk.org/r/2591/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
4cea902020 |
Corrected comment on stasis_cache_get
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390585 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
|
94d8d0468f |
Remove remaining traces of remove_on_pull from hooks and hook APIs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390550 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
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79022c0f88 |
Give the AST_BRIDGE_HOOK_REMOVE_ON_PULL a legitimate value.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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4789c3fb0c |
Change the remove_on_pull flag on ast_bridge_hook to be a set of flags.
This change is used to make bridge hook removal more generic. This way, depending on the circumstance, the appropriate bridge hooks may be removed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390510 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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cc97274d3b |
Corrected the docs on ast_manager_event_blob_create
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390398 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6d805dc04b |
Correct autoconf script for finding UUID support.
The library that provides UUID support varies greatly from system to system. On most Linux distros, it's in libuuid. On OpenBSD, it's in libe2fs-uuid. On OS X, it is in libsystem. This patch plays hide-and-seek with UUID support, looking for it in the three places we know about. It also corrects the Makefile so that it uses the configured library name and include path. (closes issue ASTERISK-21816) Reported by: Brad Latus (snuffy) Tested by: Brad Latus (snuffy) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390352 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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680765d452 |
Remove ast_channel_bridge() and associated code called only by it.
* Added some more BUGBUG notes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390291 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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ccc8cc5346 |
Fixup hold/unhold with attended and blind transfers.
* DTMF attended and blind transfers have hold/unhold behavior restored. * External attended and blind transfers unhold the transfered party when the transfer is initiated. * Made prohibit blind transferring a bridge marked as masquerade only. (ConfBridge bridges) * Made running an application or playing a file inside a bridge post the hold/unhold messages if MOH is requested. Review: https://reviewboard.asterisk.org/r/2574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a1494300c9 |
Replace ast_manager_publish_message() with a more useful version.
It's much easier to just create a blob of the message. Convert some AMI events to use it. Review: https://reviewboard.asterisk.org/r/2577/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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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 |
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e1bff7958a |
Add snapshot cache that indexes by channel name
This adds a new channel snapshot cache in parallel to the existing cache; the difference being that it indexes the channel snapshots by channel name instead of channel uniqueid. Review: https://reviewboard.asterisk.org/r/2576 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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d81c846724 |
Avoid unnecessary cleanups during immediate shutdown
This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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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 |
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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 |
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154fbf8cae |
Split Hold event into Hold/Unhold, and move it into core.
(closes issue ASTERISK-21487) Review: https://reviewboard.asterisk.org/r/2565/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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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 |
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32a86f902a |
stasis-http: Provide a response body for 201 created responses
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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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
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13 years ago |
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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
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13 years ago |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |