How do I connect to the call/app from SIP or Softphone?

[vcidst]

Hello, thank you for this example. I have the Asterisk connected to a SIP Trunk and also a Voip Softphone connected to the Asterisk. I am able to start the server and see the connection logs on the node server as well as asterisk.

However how do I connect to this application from either SIP or Softphone? My expectation is that either I can dial out a particular extension to connect to this application or a local phone will receive an outbound call from the application.

Below are some logs that I see on the node server and Asterisk,

node server

❯ bin/ari-transcriber -a "http://asterisk-dev:8088" --format=slin16 'Local/1234'
Creating ARI Controller to Asterisk instance http://asterisk-dev:8088
Starting audio listener on 127.0.0.1:9999
Starting speech provider
Creating Bridge and Channels
server listening 127.0.0.1:9999
[
  'This API is using a deprecated version of Swagger!  Please see http://github.com/wordnik/swagger-core/wiki for more info'
]
Processing

and the asterisk server,

 Creating Stasis app 'externalMedia'
  == WebSocket connection from '100.103.250.60:63597' for protocol '' accepted using version '13'
    -- Called 1234
    -- Executing [1234@default:1] playback("Local/1234@default-00000002;2", "transfer,skip")
    -- Auto fallthrough, channel 'Local/1234@default-00000002;2' status is 'UNKNOWN'
       > 0x7f6fe801d130 -- Strict RTP learning after remote address set to: 127.0.0.1:9999
    -- Called 127.0.0.1:9999
    -- UnicastRTP/127.0.0.1:9999-0x7f6fe8011310 answered
       > Launching Stasis(externalMedia) on UnicastRTP/127.0.0.1:9999-0x7f6fe8011310
    -- Channel UnicastRTP/127.0.0.1:9999-0x7f6fe8011310 joined 'simple_bridge' stasis-bridge <febb98ec-c3b7-4203-8595-715644eefa72>