feat: add initial Hold Slayer AI telephony gateway implementation

Complete project scaffolding and core implementation of an AI-powered
telephony system that calls companies, navigates IVR menus, waits on
hold, and transfers to the user when a human answers.

Key components:
- FastAPI server with REST API, WebSocket, and MCP (SSE) interfaces
- SIP/VoIP call management via PJSUA2 with RTP audio streaming
- LLM-powered IVR navigation using OpenAI/Anthropic with tool calling
- Hold detection service combining audio analysis and silence detection
- Real-time STT (Whisper/Deepgram) and TTS (OpenAI/Piper) pipelines
- Call recording with per-channel and mixed audio capture
- Event bus (asyncio pub/sub) for real-time client updates
- Web dashboard with live call monitoring
- SQLite persistence via SQLAlchemy with call history and analytics
- Notification support (email, SMS, webhook, desktop)
- Docker Compose deployment with Opal VoIP and Opal Media containers
- Comprehensive test suite with unit, integration, and E2E tests
- Simplified .gitignore and full project documentation in README
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# Configuration
All configuration is via environment variables, loaded through Pydantic Settings. Copy `.env.example` to `.env` and edit.
## Environment Variables
### SIP Trunk
| Variable | Description | Default | Required |
|----------|-------------|---------|----------|
| `SIP_TRUNK_HOST` | Your SIP provider hostname | — | Yes |
| `SIP_TRUNK_PORT` | SIP signaling port | `5060` | No |
| `SIP_TRUNK_USERNAME` | SIP auth username | — | Yes |
| `SIP_TRUNK_PASSWORD` | SIP auth password | — | Yes |
| `SIP_TRUNK_DID` | Your phone number (E.164) | — | Yes |
| `SIP_TRUNK_TRANSPORT` | Transport protocol (`udp`, `tcp`, `tls`) | `udp` | No |
### Gateway
| Variable | Description | Default | Required |
|----------|-------------|---------|----------|
| `GATEWAY_SIP_PORT` | Port for device SIP registration | `5080` | No |
| `GATEWAY_RTP_PORT_MIN` | Minimum RTP port | `10000` | No |
| `GATEWAY_RTP_PORT_MAX` | Maximum RTP port | `20000` | No |
| `GATEWAY_HOST` | Bind address | `0.0.0.0` | No |
### LLM
| Variable | Description | Default | Required |
|----------|-------------|---------|----------|
| `LLM_BASE_URL` | OpenAI-compatible API endpoint | `http://localhost:11434/v1` | No |
| `LLM_MODEL` | Model name for IVR analysis | `llama3` | No |
| `LLM_API_KEY` | API key (if required) | `not-needed` | No |
| `LLM_TIMEOUT` | Request timeout in seconds | `30.0` | No |
| `LLM_MAX_TOKENS` | Max tokens per response | `1024` | No |
| `LLM_TEMPERATURE` | Sampling temperature | `0.3` | No |
### Speech-to-Text
| Variable | Description | Default | Required |
|----------|-------------|---------|----------|
| `SPEACHES_URL` | Speaches/Whisper STT endpoint | `http://localhost:22070` | No |
| `SPEACHES_MODEL` | Whisper model name | `whisper-large-v3` | No |
### Database
| Variable | Description | Default | Required |
|----------|-------------|---------|----------|
| `DATABASE_URL` | PostgreSQL or SQLite connection string | `sqlite+aiosqlite:///./hold_slayer.db` | No |
### Notifications
| Variable | Description | Default | Required |
|----------|-------------|---------|----------|
| `NOTIFY_SMS_NUMBER` | Phone number for SMS alerts (E.164) | — | No |
### Audio Classifier
| Variable | Description | Default | Required |
|----------|-------------|---------|----------|
| `CLASSIFIER_WINDOW_SECONDS` | Audio window size for classification | `3.0` | No |
| `CLASSIFIER_SILENCE_THRESHOLD` | RMS below this = silence | `0.85` | No |
| `CLASSIFIER_MUSIC_THRESHOLD` | Spectral flatness below this = music | `0.7` | No |
| `CLASSIFIER_SPEECH_THRESHOLD` | Spectral flatness above this = speech | `0.6` | No |
### Hold Slayer
| Variable | Description | Default | Required |
|----------|-------------|---------|----------|
| `MAX_HOLD_TIME` | Maximum seconds to wait on hold | `7200` | No |
| `HOLD_CHECK_INTERVAL` | Seconds between audio checks | `2.0` | No |
| `DEFAULT_TRANSFER_DEVICE` | Device to transfer to | `sip_phone` | No |
### Recording
| Variable | Description | Default | Required |
|----------|-------------|---------|----------|
| `RECORDING_DIR` | Directory for WAV recordings | `recordings` | No |
| `RECORDING_MAX_SECONDS` | Maximum recording duration | `7200` | No |
| `RECORDING_SAMPLE_RATE` | Audio sample rate | `16000` | No |
## Settings Architecture
Configuration is managed by Pydantic Settings in `config.py`:
```python
from config import get_settings
settings = get_settings()
settings.sip_trunk_host # "sip.provider.com"
settings.llm.base_url # "http://localhost:11434/v1"
settings.llm.model # "llama3"
settings.speaches_url # "http://localhost:22070"
settings.database_url # "sqlite+aiosqlite:///./hold_slayer.db"
```
LLM settings are nested under `settings.llm` as a `LLMSettings` sub-model.
## Deployment
### Development
```bash
# 1. Clone and install
git clone <repo-url>
cd hold-slayer
python -m venv .venv
source .venv/bin/activate
pip install -e ".[dev]"
# 2. Configure
cp .env.example .env
# Edit .env
# 3. Start Ollama (for LLM)
ollama serve
ollama pull llama3
# 4. Start Speaches (for STT)
docker run -p 22070:8000 ghcr.io/speaches-ai/speaches
# 5. Run
uvicorn main:app --host 0.0.0.0 --port 8000 --reload
```
### Production
```bash
# Use PostgreSQL instead of SQLite
DATABASE_URL=postgresql+asyncpg://user:pass@localhost/hold_slayer
# Use vLLM for faster inference
LLM_BASE_URL=http://localhost:8000/v1
LLM_MODEL=meta-llama/Llama-3-8B-Instruct
# Run with multiple workers (note: each worker is independent)
uvicorn main:app --host 0.0.0.0 --port 8000 --workers 1
```
Note: Hold Slayer is designed as a single-process application. Multiple workers would each have their own SIP engine and call state. For high availability, run behind a load balancer with sticky sessions.
### Docker
```dockerfile
FROM python:3.13-slim
# Install system dependencies for PJSUA2 and Sippy
RUN apt-get update && apt-get install -y \
build-essential \
libpjproject-dev \
&& rm -rf /var/lib/apt/lists/*
WORKDIR /app
COPY . .
RUN pip install -e .
EXPOSE 8000 5080/udp 10000-20000/udp
CMD ["uvicorn", "main:app", "--host", "0.0.0.0", "--port", "8000"]
```
Port mapping:
- `8000` — HTTP API + WebSocket + MCP
- `5080/udp` — SIP device registration
- `10000-20000/udp` — RTP media ports