On Sun, Jun 15, 2014 at 9:24 PM, Krandon <krandon.bruse@xxxxxxxxx> wrote:
Hello Asterisk friends,I am currently interfacing with Asterisk through ARI and loving the experience so far. I have successfully originated calls and dumped them into my Stasis app. I am trying to figure out what the best way is to send a channel into an Application. The current architecture for /channels/{id}/play works well for the majority of my app, but I am running into a block figuring out how to interact with Asterisk dialplan applications.To give an example - I submit an originate to go to SIP/vendor/phoneNumber - with the other leg going to App: myStasisApp, {"soundFile":"blah"}. That works fine (with the proper quote escaping). Now my Stasis app has received the channelID to which we can do a lot of neat stuff. Say I play a sound to the user but then want to call the app WaitForSilence. What's the best way to do this? I may be misinterpreting the intended use of both Stasis and ARI - but I am curious to see what your thoughts are.Also, for the stasis app to get a list of arguments, I am passing it through as JSON. So far that is working fine - but I wanted to see if there was a better way to get a list/array of app args to Stasis.Forgive me if there is an easy solution - through digging and poking the last few days, I have not been able to find the intended use case or even a use case.
Well, the solution for this just got added into the Asterisk 12 branch, and so it hasn't made it into a release yet. It should be coming soon in Asterisk 12.4.0.
The TALK_DETECT [1] function enables AMI/ARI events [2] [3] [4] [5] on a channel, such that a connected ARI application receives notifications over the WebSocket when a person starts/stops talking. This lets you asynchronously 'know' when both talking/silence has occurred - obviating the need for the WaitForSilence/WaitForNoise dialplan applications. Plus, because it is asynchronous, if you decide you don't *want* to wait for silence, you don't have to!
With a bit of manipulation, you could also construct AMD from this as well - but I'll admit that's a bit more challenging. I'd be interested in people's experiences with attempting to do that, and if an asynchronous "IS_HUMAN" detection function is needed or not.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_TALK_DETECT
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_ChannelTalkingStart
[3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_ChannelTalkingStop
[4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+REST+Data+Models#Asterisk12RESTDataModels-ChannelTalkingStarted
[5] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+REST+Data+Models#Asterisk12RESTDataModels-ChannelTalkingFinished
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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