Skill Asterisk

OpenAI Realtime ↔ Asterisk Bridge (Node.js + JsSIP)

This skill ships a small Node.js app that:

• Registers a WebRTC SIP endpoint to Asterisk via SIP-over-WebSocket (JsSIP).
• Places an outgoing call (by phonebook name or number).
• Streams call audio → OpenAI Realtime input.
• Streams OpenAI Realtime audio output → the SIP call.

One-time setup

1. Ensure Asterisk is configured for WebRTC + SIP-over-WebSocket (PJSIP WSS transport, DTLS-SRTP, ICE).
2. Install Node dependencies:
   • cd scripts && npm install

Configure

Edit scripts/bridge.js and hardcode at the top:

• OPENAI_API_KEY
• SIP_URI (example: sip:1234@test.example.com)
• SIP_USER (example: 1234)
• SIP_PASS (example: 12345)

Also edit:

• phonebook.txt (name ↔ number ↔ description)
• Optional SIP_WS override (defaults to wss://<domain-from-SIP_URI>/ws)

Phonebook format

Edit phonebook.txt (freeform lines). Supported format is comma-separated:

• Name, Phone Number, Description (optional)

Run

• List phonebook entries:
  • node scripts/bridge.js --list
• Call by name:
  • node scripts/bridge.js --to Alice --prompt "Say hello and ask about the report."
• Call by number/extension:
  • node scripts/bridge.js --to 1234 --prompt "Hello!"

Notes

• This requires network access (OpenAI + your Asterisk WSS endpoint).
• @roamhq/wrtc is a native module; you may need build tooling (Python + make + C/C++ toolchain) for npm install.