voicegsmtest end-to-end GSM tester

Dialer · Mic softphone · AI agent · Campaign — all against the GSM gateway.

AIchecking AMIdisconnected WSconnecting

Browser softphone

Register your browser as a PJSIP WebRTC endpoint. Mic → WebRTC → Asterisk → GSM trunk → phone. Dialpad strips a leading 9 via the [from-webphone] dialplan.

SIP not registered
Call idle

Single call (server-originated)

Trigger one call via AMI. Pick a mode below — AI bridges audio to OpenAI Realtime; plain dial just rings the number through the GSM trunk. 


    


    
    

      
Campaign (bulk sequential)

      

        Upload a list of numbers, pick a mode, hit Start. Calls are dialed one at a time
        with the configured delay between each. Live progress streams below.
      

      

        
        
        
        
      

      
      

        
        
      

      


      
Campaigns

      


      
Active contacts (selected campaign)

      

    


    
    

      
Live events

      

        AMI events + AudioSocket sessions + AI transcripts + campaign progress.
        Streamed via WebSocket from /events.