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99 lines (72 loc) · 2.46 KB
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# ====================================
# Claude Phone Configuration
# ====================================
# Copy this to .env and fill in your values
# ====================================
# Network Configuration
# ====================================
# Your server's LAN IP (CRITICAL for RTP audio)
# This IP goes into SDP packets so 3CX knows where to send audio
EXTERNAL_IP=10.0.0.100
# ====================================
# Drachtio Configuration
# ====================================
DRACHTIO_HOST=127.0.0.1
DRACHTIO_PORT=9022
DRACHTIO_SECRET=cymru
# SIP port for Contact header (5070 for 3CX SBC mode, 5060 for direct registration)
# This MUST match the port in docker-compose.yml drachtio command
DRACHTIO_SIP_PORT=5060
# ====================================
# FreeSWITCH Configuration
# ====================================
FREESWITCH_HOST=127.0.0.1
FREESWITCH_PORT=8021
FREESWITCH_SECRET=JambonzR0ck$
# ====================================
# 3CX / SIP Configuration
# ====================================
# Your 3CX server hostname or IP
SIP_DOMAIN=your-3cx-server.3cx.us
SIP_REGISTRAR=10.0.0.50
# Default extension for outbound calls
SIP_EXTENSION=9000
SIP_AUTH_ID=your-auth-id
SIP_PASSWORD=your-password
# Default caller ID for outbound PSTN calls
DEFAULT_CALLER_ID=+15551234567
# ====================================
# Claude API Server
# ====================================
# URL to your Claude API server (runs on your API server with Claude Max)
CLAUDE_API_URL=http://10.0.0.200:3333
# ====================================
# ElevenLabs TTS
# ====================================
ELEVENLABS_API_KEY=your-elevenlabs-api-key
# Default voice ID (if not specified per-device)
ELEVENLABS_VOICE_ID=your-default-voice-id
# ====================================
# OpenAI (Whisper STT)
# ====================================
OPENAI_API_KEY=your-openai-api-key
# ====================================
# Application Settings
# ====================================
# HTTP server port for API and audio file serving
HTTP_PORT=3000
# WebSocket port for audio streaming from FreeSWITCH
WS_PORT=3001
# Audio files directory
AUDIO_DIR=/app/audio
# ====================================
# Outbound Call Settings
# ====================================
# Maximum conversation turns for outbound calls
MAX_CONVERSATION_TURNS=10
# Ring timeout in seconds
OUTBOUND_RING_TIMEOUT=30
# ====================================
# Optional: Debug
# ====================================
# DEBUG=voice-app:*