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1.4.30
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${ noResults }
24131 Commits (b193c2873d27e15dfec521c7fef10411e8d1f5c8)
Author | SHA1 | Message | Date |
---|---|---|---|
|
b193c2873d |
Handle hangup logic in the Stasis message bus and consumers of Stasis messages
This patch does the following: * It adds a new soft hangup flag AST_SOFTHANGUP_HANGUP_EXEC that is set when a channel is executing dialplan hangup logic, i.e., the 'h' extension or a hangup handler. Stasis messages now also convey the soft hangup flag so consumers of the messages can know when a channel is executing said hangup logic. * It adds a new channel flag, AST_FLAG_DEAD, which is set when a channel is well and truly dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs, and other consumers of Stasis have been updated to look for this flag to know when the channel should by lying six feet under. * The CDR engine has been updated to better handle a channel entering and leaving a bridge. Previously, a new CDR was automatically created when a channel left a bridge and put into the 'Pending' state; however, this way of handling CDRs made it difficult for the 'endbeforehexten' logic to work correctly - there was always a new CDR waiting in the hangup logic and, even if 'ended', wouldn't be the CDR people wanted to inspect in the hangup routine. This patch completely removes the Pending state and instead defers creation of the new CDR until it gets a new message that requires a new CDR. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393777 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
c54b26a18c |
ARI: return a 503 if Asterisk isn't fully booted
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393768 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
cc3478d2e8 |
Print error details when set nonblock fails
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
01c21c7aea |
Document MissingParams error message for /ari/events
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393749 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
d0a55fa52d |
Refactor RTCP events over to Stasis; associate with channels
This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
d789681eaf |
OneTouchRecord: Add function defined earlier: ast_bridge_features_do()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393729 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
4123b27f6f |
Remove parkinglot from the channel snapshot
Legacy channel drivers often include the ability to set a default parking lot on an endpoint basis; when channels are created for that endpoint, they inherit the parkinglot option. Parking used to use this option more frequently; while it is still supported, other options (such as using channel variables or creation of a custom parkinglot) are supported. More importantly, conveying the parkinglot information through a channel snapshot isn't terribly useful - it is rarely (if ever) changed on a channel and some consumers of channel snapshots, such as ARI, will never use the information. (closes issue ASTERISK-21968) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393716 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
93ed5ef0ff |
res_parking: Replace Parker snapshots with ParkerDialString
This process also involved a large amount of rework regarding how to redial the Parker when a channel leaves a parking lot due to timeout. An attended transfer channel variable has been added to attended transfers to extensions that will eventually park (but haven't at the time of transfer) as well. This resolves one of the two BUGBUG comments remaining in res_parking. (issues ASTERISK-21877) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2638/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
fb03bf9b39 |
Fix int width problem for 32-bit... again
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
951c87634f |
Fix int width problem for 32-bit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
04cde027d4 |
Fix utils directory breakage.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
415b79dec9 |
Add BUGBUG note for ASTERISK-22009
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393633 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 |
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c841c34aae |
chan_dahdi: Fix segfault reloading chan_dahdi when round robin is used.
* Clear round_robin[] in dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 393627 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 393628 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393629 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
0227e00eb3 |
OneTouchRecord: Make so Monitor/MixMonitor can be toggled/started/stopped.
The OneTouchRecord feature has historically been a toggle. This patch adds the ability to make the OneTouchRecord hook optionally start/stop recording only. If OneTouchRecord is already doing what is requested then only the invoker hears the courtesy tone and/or start/stop recording message. The new feature is written so we could easily add explicit start/stop recording DTMF hooks for Monitor and MixMonitor. The majority of the changes in bridge_builtin_features.c is a refactoring of the OneTouchRecord code (Monitor and MixMonitor versions) so it is easy to direct the toggle/start/stop functionality. Review: https://reviewboard.asterisk.org/r/2655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393612 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
ad5dc3c159 |
Move when bridge channel enter is published so it does not interrupt the thought of some lines of code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393601 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
b96d8cbc78 |
Fix some indentation in stasis_config.c.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393600 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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a981de9adb |
Fix some bugs in CDRs; add some CLI commands to help debugging
This patch fixes a few minor bugs and one major one: the CDR by bridge container was less than helpful. The mechanism previously used to try and find all of the CDRs in a particular bridge ended up missing CDRs, resulting in incorrect records. When looking up CDRs in a bridge, we now just bite the bullet and do a selection across all existing CDRs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393599 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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569f5f2117 |
Let Stasis load itself with default values
While a Stasis configuration file is nice, it shouldn't be mandatory. We can carry on with default values. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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0f725bd5d9 |
Publish a bridge enter before pulling on a push-and-swap operation.
Prior to this patch, the order of procedures on a bridge push was * Add new bridge channel to bridge's array. * Pull the swap channel out of the bridge * Publish a bridge enter event. The problem is that when the swap channel was pulled from the bridge, a bridge leave event would be published. The bridge snapshot published during the bridge leave showed the new channel that had been added to the bridge, but there had been no bridge enter event for that channel. The fix provided here was to change the order a bit * Add new bridge channel to bridge's array. * Publish bridge enter event. * Pull the swap channel out of the bridge. This makes it so that the bridge snapshots during the stasis events are accurate. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
dbc588b02f |
Fix load errors related to the new ari_model_validators.
The Asterisk strategy of loading modules with RTLD_LAZY to extract metadata from the module works well enough, until you try to take the address of a function. If a module takes the address of a function, that function needs to be resolved at load time. That kinda defeats RTLD_LAZY. This patch adds some ari_validator_{id}_fn() wrapper functions for safely getting the function pointer from a different module. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
ef032842f1 |
Violating the margins to make menuconfig happy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393561 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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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 |
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c4adaf9106 |
Configuration for Stasis threadpool
The appropriate settings for the Stasis threadpool is very system specific, depending upon both workload and system configuration. This patch adds a stasis.conf file which can be used to configure the key attributes of the threadpool for the Stasis message bus. (closes issue ASTERISK-21280) Review: https://reviewboard.asterisk.org/r/2651/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
9ba976b19c |
ARI authentication.
This patch adds authentication support to ARI. Two authentication methods are supported. The first is HTTP Basic authentication, as specified in RFC 2617[1]. The second is by simply passing the username and password as an ?api_key query parameter (which allows swagger-ui[2] to authenticate more easily). ARI usernames and passwords are configured in the ari.conf file (formerly known as stasis_http.conf). The user may be set to `read_only`, which will prohibit the user from issuing POST, DELETE, etc. Also, the user's password may be specified in either plaintext, or encrypted using the crypt() function. Several other notes about the patch. * A few command line commands for seeing ARI config and status were also added. * The configuration parsing grew big enough that I extracted it to its own file. [1]: http://www.ietf.org/rfc/rfc2617.txt [2]: https://github.com/wordnik/swagger-ui (closes issue ASTERISK-21277) Review: https://reviewboard.asterisk.org/r/2649/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
c9a3d4562d |
Update events to use Swagger 1.3 subtyping, and related aftermath
This patch started with the simple idea of changing the /events data model to be more sane. The original model would send out events like: { "stasis_start": { "args": [], "channel": { ... } } } The event discriminator was the field name instead of being a value in the object, due to limitations in how Swagger 1.1 could model objects. While technically sufficient in communicating event information, it was really difficult to deal with in terms of client side JSON handling. This patch takes advantage of a proposed extension[1] to Swagger which allows type variance through the use of a discriminator field. This had a domino effect that made this a surprisingly large patch. [1]: https://groups.google.com/d/msg/wordnik-api/EC3rGajE0os/ey_5dBI_jWcJ In changing the models, I also had to change the swagger_model.py processor so it can handle the type discriminator and subtyping. I took that a big step forward, and using that information to generate an ari_model module, which can validate a JSON object against the Swagger model. The REST and WebSocket generators were changed to take advantage of the validators. If compiled with AST_DEVMODE enabled, JSON objects that don't match their corresponding models will not be sent out. For REST API calls, a 500 Internal Server response is sent. For WebSockets, the invalid JSON message is replaced with an error message. Since this took over about half of the job of the existing JSON generators, and the .to_json virtual function on messages took over the other half, I reluctantly removed the generators. The validators turned up all sorts of errors and inconsistencies in our data models, and the code. These were cleaned up, with checks in the code generator avoid some of the consistency problems in the future. * The model for a channel snapshot was trimmed down to match the information sent via AMI. Many of the field being sent were not useful in the general case. * The model for a bridge snapshot was updated to be more consistent with the other ARI models. Another impact of introducing subtyping was that the swagger-codegen documentation generator was insufficient (at least until it catches up with Swagger 1.2). I wanted it to be easier to generate docs for the API anyways, so I ported the wiki pages to use the Asterisk Swagger generator. In the process, I was able to clean up many of the model links, which would occasionally give inconsistent results on the wiki. I also added error responses to the wiki docs, making the wiki documentation more complete. Finally, since Stasis-HTTP will now be named Asterisk REST Interface (ARI), any new functions and files I created carry the ari_ prefix. I changed a few stasis_http references to ari where it was non-intrusive and made sense. (closes issue ASTERISK-21885) Review: https://reviewboard.asterisk.org/r/2639/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
dcf03554a0 |
Shuffle RESTful URL's around.
This patch moves the RESTful URL's around to more appropriate locations for release. The /stasis URL's are moved to /ari, since Asterisk REST Interface was a more appropriate name than Stasis-HTTP. (Most of the code still has stasis_http references, but they will be cleaned up after there are no more outstanding branches that would have merge conflicts with such a change). A larger change was moving the ARI events WebSocket off of the shared /ws URL to its permanent home on /ari/events. The Swagger code generator was extended to handle "upgrade: websocket" and "websocketProtocol:" attributes on an operation. The WebSocket module was modified to better handle WebSocket servers that have a single registered protocol handler. If a client connections does not specify the Sec-WebSocket-Protocol header, and the server has a single protocol handler registered, the WebSocket server will go ahead and accept the client for that subprotocol. (closes issue ASTERISK-21857) Review: https://reviewboard.asterisk.org/r/2621/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
85ba063329 |
Add a SystemName field to all AMI events.
This only gets sent out if configured in asterisk.conf (closes issue ASTERISK-21494) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393508 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 |
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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 |
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96fed373e9 |
Fix chan_gtalk.c compile error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393485 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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d98c19f947 |
Add pjproject dependency to res_sip_notify
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393484 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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69242ad317 |
Remove unused blind transfer publication structure.
I ended up using a bridge blob, so this structure was unused. Keeping it in the header would just cause confusion. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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05a16729cb |
Stasis - Refactor AOC Events
Refactored the AMI events in AOC onto Stasis-Core. The ast_aoc_manager_event function now publishes a channel snapshot, along with a JSON blob describing the advice of charge. A "to_ami" handler has also been added that converts the channel snapshot and AOC event data back into the appropriate data structure for use with AMI. (closes issue ASTERISK-21472) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2643/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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a25a630659 |
New SIP Channel driver: Always Auth Reject
If no matching endpoint is found for the incoming request Asterisk will respond with a 401 Unauthorized (rejecting the request), but will first challenge if no authorization creditials are given. Changes also included moving ACL options into a new global 'security' configuration section in res_sip.conf. (closes issue ASTERISK-21433) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2554/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393442 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
de206baa99 |
Fix transfer AMI event parameter naming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393429 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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6e3a5d2d48 |
Add CEL unit tests and do some cleanup
This adds several unit tests for CEL functionality and provides the requisite framework for creating additional unit tests. This also cleans up some reference leaks that were occurring in Stasis-Core message callback code. Review: https://reviewboard.asterisk.org/r/2646/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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f7624718f8 |
Fix issue with inability to cancell call transfer made by on-sceen menus.
|
12 years ago |
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51c54ddf8d |
ast_tls_cert: don't recreate generated files
Don't regenrate cat.cfg, ca.crt and ca.key if they were already created on a previous run. (closes issue ASTERISK-21932) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393383 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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5456794b66 |
New SIP Channel Driver - Add CLI/AMI initiated NOTIFY requests
Added the ability to send unsolicited NOTIFY requests to a particular endpoint with a configured payload. Added both CLI and AMI support. For a given endpoint, this module will iterate over all its contacts sending the appropriate NOTIFY request to each. (closes issue ASTERISK-21436) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2623/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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3841520a6e |
Prevent crash during synchronous AMI origination by ref bumping returned channel
The originate APIs allow callers to provide a pointer to a channel that will point to the originated channel if the function call succeeds. This is used by AMI to provide channel information when the originate is performed synchronously. Unfortunately, if the originate fails in certain ways, the outbound channel is already disposed of during the dialing itself. This results in the channel being improperly dereferenced by the internal originate function in pbx.c. This patch ref bumps the channel to prevent this from occurring. Callers must now unlock and unref the channel (which is more in line with general channel management guidelines anyway). This only affects manager, as it is the only consumer of this API function that actually passes in a channel pointer. Review: https://reviewboard.asterisk.org/r/2617/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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f820d24db1 |
ARI: Implement channel hold/unhold.
This puts the channel on hold (rather than queueing a frame from the channel). (closes issue ASTERISK-21619) Review: https://reviewboard.asterisk.org/r/2647/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393332 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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f41faf0b7d |
ARI: Implement channel dial.
This creates a new outbound channel, and bridges it to a channel already in the Stasis application. (closes issue ASTERISK-21620) Review: https://reviewboard.asterisk.org/r/2634/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393326 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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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 |