Is it possible to have a Stasis call controlled through ARI and use the Dial to connect the channel's audio to a websocket server?

Asterisk Asterisk APIs
1

We have an older application using AMI. It can send the audio to our websocket server.
Kudos to those who worked on this new feature!!!

Action: AGI
ActionID: C59
Channel: PJSIP/100-00000003
CommandID: C59
Command: EXEC Dial WebSocket/1/c(slin16)v(urlparamname1=urlparamvalue1,urlparamname2=urlparamvalue2),30

  1. Is it possible for the Dial to be used for this as well?
  2. If so, would the channel value be?

WebSocket/1/c(slin16)v(urlparamname1=urlparamvalue1,urlparamname2=urlparamvalue2)

I haven’t attempted a Dial via ARI before, but ran some tests by hand.

I tried simulating manually via curl requests and it wasn’t working for me. Then, I decided to verify I’m passing everything correctly via the curl by dialing a phone and also ran into problems.

Guessing I’m misunderstanding how to pass the parameters via ARI?

AMI, the following does call my phone 1005 on the PJSIP endpoint 1005

Action: AGI
ActionID: C60
Channel: PJSIP/100-00000003
CommandID: C60
Command: EXEC Dial PJSIP/1005@1005,30

However, when I try this via curl/ARI it doesn’t work (yes, I have the call in Stasis)

curl -v -u user:password -X POST “http://myip:myport/ari/channels/1756934869.22/dial?channel=PJSIP/1005@1005&timeout=30”

For comparison, I can start playing music on hold using a similar request
curl -v -u user:password -X POST “http://myip:myport/ari/channels/1756934869.22/moh?mohClass=1”
3) Any suggestions on what I am doing incorrectly via the ARI request? Is it not possible to use ARI to dial a number?

From the API, it looks like it wants to dial an existing channel so I tried that and it also doesn’t work.
curl -v -u user:password -X POST “http://myip:myport/ari/channels/1756934869.22/dial?channel=1234567890.46&timeout=30”

Any thoughts/suggestions?

Dan

2

I’m checking the dial syntax but what kind of responses are you getting from the curl calls?

3

This doesn’t do what you think it does. It is not the equivalent of the Dial application. It is for dialing channels created using the create[1] route[2].

You can use that to place an outgoing call, or use a normal originate[3].

ARI gives you the fundamentals to do things, not the higher level concepts (like have a channel dial another channel and connect them together all in one call).

[1] Channels - Asterisk Documentation

[2] Asterisk 14 ARI: Create, Bridge, Dial. ⋆ Asterisk

[3] Channels - Asterisk Documentation

4

And it’s “caller=” not “channel=”

From a websocket perspective, this would be the flow…

  • Receive StasisStart for an incoming PJSIP channel.
  • Call channels/create for the WebSocket channel.
  • Call channels/<websocket_channel_id>/dial?caller=<pjsip_channel_id>
  • Watch for the Dial status ANSWER and bridge the two channels.

The [asterisk-websocket-examples](GitHub - asterisk/asterisk-websocket-examples: Examples for using Asterisk ARI and Media WebSockets) repo has examples.

5

Thank you

6

Thank you George. You did an awesome job on this feature.
This is exactly what I’m looking for.

(Clearly I need another cup of coffee to wakeup …caller instead of channel.)

7

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