# asterisk-ai-voice-agent — infrastructure config.
# Copy to config.yaml and fill in. Personas live in personas.yaml.

listen:
  host: 127.0.0.1        # keep AudioSocket loopback-only
  audiosocket_port: 9092 # Asterisk AudioSocket() dials this
  register_port: 9091    # tiny HTTP endpoint the dialplan pre-registers persona to
  # websocket_port: 9093 # optional: chan_websocket (Asterisk 20.18+/22.8+).
                         # Needs control_message_format = json in chan_websocket.conf,
                         # and the dial string must carry the call id:
                         #   Dial(WebSocket/conn1/c(slin),v(uuid=${CALLUUID}))
                         # Leave commented out to run AudioSocket only.

providers:
  anthropic:
    api_key: ""          # required for the default LLM
  openai:
    api_key: ""          # Whisper STT (the default)
  elevenlabs:
    api_key: ""          # optional: Scribe STT and/or cloud TTS (Piper is the local default)
  sixtydb:
    api_key: ""          # optional: 60db final transcription
  piper:
    voices_dir: ./voices # local Piper .onnx voices (download_voices.sh)
    default_voice: en_US-amy-medium

tools:
  # Where LLM tool_use calls and end-of-call transcripts are POSTed.
  # Leave blank to disable tool calling entirely.
  webhook_url: ""
  webhook_auth: ""       # optional value sent as `Authorization` header

limits:
  max_concurrent_calls: 2   # each call ~150 MB during synthesis bursts
  max_call_seconds: 900     # hard safety cap per call
  stt_timeout_s: 30
  llm_timeout_s: 30
  tts_timeout_s: 15