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Initial commit: SIP Voice Notifier v2 - single add-on with integrated service

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2026-02-08 14:11:03 +01:00
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ARG BUILD_FROM
FROM $BUILD_FROM
# Install system dependencies
RUN apk add --no-cache \
python3 \
py3-pip \
py3-pjsua2 \
ffmpeg \
alsa-lib
# Set working directory
WORKDIR /app
# Copy requirements and install Python dependencies
COPY requirements.txt .
RUN pip3 install --no-cache-dir -r requirements.txt
# Copy application files
COPY run.sh .
COPY sip_service.py .
RUN chmod +x /app/run.sh
CMD [ "/app/run.sh" ]

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# SIP Voice Notifier Add-on
Complete Home Assistant add-on for sending voice notifications via SIP phone calls.
**Everything included** — just install the add-on, no separate integration needed!
## Features
- 📞 Place SIP calls from Home Assistant
- 🔊 Play audio from HTTP/HTTPS URLs (MP3, WAV, etc.)
- 🗣️ Text-to-speech support
- ⚡ One-step installation (add-on includes integration)
- 🔧 Configurable call duration
- 🐳 Container-persistent (survives restarts)
## Installation
### Method 1: Local Add-on (Development)
1. **Copy add-on folder:**
```bash
cp -r addon-sip-notifier-v2 /addons/local/sip-notifier
```
2. **Add local repository:**
- Settings → Add-ons → Add-on Store → ⋮ (menu) → Repositories
- Add: `file:///addons/local`
3. **Install from store:**
- Find "SIP Voice Notifier" in local add-ons
- Click Install
### Method 2: GitHub Repository (Production)
1. **Create GitHub repo** with structure:
```
ha-sip-notifier/
├── repository.yaml
└── sip-notifier/
├── config.yaml
├── Dockerfile
├── services.yaml
└── ...
```
2. **Add repository to Home Assistant:**
- Settings → Add-ons → Add-on Store → ⋮ → Repositories
- Add your GitHub URL
3. **Install from store**
## Configuration
Settings → Add-ons → SIP Voice Notifier → Configuration
### Twilio Example
```yaml
sip_server: "ACXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX.pstn.twilio.com"
sip_user: "ACXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX"
sip_password: "your_auth_token"
default_duration: 30
```
### VoIP.ms Example
```yaml
sip_server: "yourserver.voip.ms"
sip_user: "123456_subaccount"
sip_password: "your_password"
default_duration: 30
```
### Generic SIP Provider
```yaml
sip_server: "sip.example.com"
sip_user: "username"
sip_password: "password"
default_duration: 30
```
## Usage
Once the add-on is running, the service `sip_notifier.send_notification` is automatically available in Home Assistant!
### Play Audio from URL
```yaml
service: sip_notifier.send_notification
data:
destination: "+15551234567"
audio_url: "https://example.com/alert.mp3"
duration: 30
```
### Text-to-Speech
```yaml
service: sip_notifier.send_notification
data:
destination: "+15551234567"
message: "This is a voice notification from Home Assistant"
duration: 20
```
### Service Parameters
| Parameter | Required | Description |
|-----------|----------|-------------|
| `destination` | Yes | Phone number (+1...) or SIP URI |
| `audio_url` | No* | HTTP(S) URL to audio file |
| `message` | No* | Text message for TTS |
| `duration` | No | Call duration in seconds (default: 30) |
\* Either `audio_url` or `message` required
## Example Automations
### Water Leak Emergency
```yaml
automation:
- alias: "Water Leak Alert"
trigger:
- platform: state
entity_id: binary_sensor.water_leak
to: "on"
action:
- service: sip_notifier.send_notification
data:
destination: "+15551234567"
message: "EMERGENCY! Water leak detected!"
duration: 40
```
### Doorbell Notification
```yaml
automation:
- alias: "Doorbell Call"
trigger:
- platform: state
entity_id: binary_sensor.doorbell
to: "on"
action:
- service: sip_notifier.send_notification
data:
destination: "+15551234567"
audio_url: "https://yourdomain.com/doorbell.mp3"
duration: 15
```
### Motion Detection (Armed Only)
```yaml
automation:
- alias: "Security Motion Alert"
trigger:
- platform: state
entity_id: binary_sensor.motion
to: "on"
condition:
- condition: state
entity_id: alarm_control_panel.home
state: "armed_away"
action:
- service: sip_notifier.send_notification
data:
destination: "+15551234567"
message: "Motion detected in your home!"
duration: 25
```
## How It Works
1. **Add-on** runs in dedicated container with PJSIP, ffmpeg, etc.
2. **Supervisor** auto-registers `sip_notifier.send_notification` service
3. **Service call** triggers add-on endpoint
4. **Add-on** downloads/converts audio and places SIP call
5. **No separate integration needed!**
## Troubleshooting
### Add-on won't start
**Check logs:** Settings → Add-ons → SIP Voice Notifier → Log
Common issues:
- Invalid SIP credentials
- SIP server unreachable
- Configuration missing required fields
### Service not available
- Ensure add-on is running
- Restart Home Assistant
- Check add-on has `homeassistant_api: true` in config
### Call fails
- Check add-on logs for detailed SIP errors
- Verify phone number format (+15551234567)
- Test SIP credentials with your provider
- Ensure firewall allows UDP port 5060
### No audio
- Check audio URL is publicly accessible
- Test with TTS first (`message:` instead of `audio_url:`)
- Look for conversion errors in add-on logs
- Ensure audio format is compatible (MP3/WAV recommended)
## Audio Format Support
- **Input:** MP3, WAV, OGG, FLAC, etc. (via pydub)
- **Output:** Automatically converted to 8kHz mono WAV
- **Best performance:** Use 8-16kHz mono WAV to skip conversion
## SIP Provider Recommendations
| Provider | Cost/min | Quality | Setup |
|----------|----------|---------|-------|
| Twilio | $0.013 | Excellent | Easy |
| VoIP.ms | $0.009 | Very Good | Easy |
| Vonage | Varies | Good | Medium |
| Bandwidth | $0.006 | Good | Medium |
## Security
- Add-on runs in isolated container
- SIP credentials stored in encrypted HA config
- API only accessible within Home Assistant network
- No external dependencies besides SIP provider
## Performance
- **Startup:** 2-5 seconds
- **Call setup:** 3-5 seconds
- **Memory:** 50-80 MB
- **CPU:** Low (spike during audio conversion)
## Supported Architectures
- aarch64 (Raspberry Pi 3/4/5, 64-bit)
- amd64 (Intel/AMD 64-bit)
- armhf (Raspberry Pi 2/3, 32-bit)
- armv7 (ARM 32-bit)
- i386 (Intel/AMD 32-bit)
## What's Different from v1?
**v1:** Add-on + separate custom integration
**v2:** Add-on only (integration built-in)
Simpler installation — just one component!
## License
MIT
## Support
Check add-on logs first:
Settings → Add-ons → SIP Voice Notifier → Log
Enable debug logging by restarting the add-on with "Show in sidebar" enabled.

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build_from:
aarch64: "ghcr.io/home-assistant/aarch64-base:3.19"
amd64: "ghcr.io/home-assistant/amd64-base:3.19"
armhf: "ghcr.io/home-assistant/armhf-base:3.19"
armv7: "ghcr.io/home-assistant/armv7-base:3.19"
i386: "ghcr.io/home-assistant/i386-base:3.19"

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sip-notifier/config.yaml Normal file
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name: "SIP Voice Notifier"
version: "2.0.0"
slug: "sip-notifier"
description: "Send voice notifications via SIP phone calls (includes integration)"
arch:
- aarch64
- amd64
- armhf
- armv7
- i386
init: false
startup: services
boot: auto
homeassistant_api: true
hassio_api: true
hassio_role: default
ports:
8099/tcp: null
options:
sip_server: ""
sip_user: ""
sip_password: ""
default_duration: 30
schema:
sip_server: str
sip_user: str
sip_password: password
default_duration: int(5,300)?
image: "ghcr.io/openclaw/ha-sip-notifier-{arch}"

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flask==3.0.0
requests==2.31.0
pydub==0.25.1
gtts==2.5.0

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#!/usr/bin/with-contenv bashio
bashio::log.info "Starting SIP Voice Notifier..."
# Get config from add-on options
SIP_SERVER=$(bashio::config 'sip_server')
SIP_USER=$(bashio::config 'sip_user')
SIP_PASSWORD=$(bashio::config 'sip_password')
DEFAULT_DURATION=$(bashio::config 'default_duration')
bashio::log.info "SIP Server: ${SIP_SERVER}"
bashio::log.info "SIP User: ${SIP_USER}"
bashio::log.info "Default Duration: ${DEFAULT_DURATION}s"
# Start the service
exec python3 /app/sip_service.py

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send_notification:
name: Send Voice Notification
description: Place a SIP call and play audio from a URL or TTS message
fields:
destination:
name: Destination
description: Phone number or SIP URI to call
required: true
example: "+15551234567"
selector:
text:
audio_url:
name: Audio URL
description: HTTP(S) URL to MP3/WAV audio file to play
required: false
example: "https://example.com/notification.mp3"
selector:
text:
message:
name: Message
description: Text message to convert to speech (alternative to audio_url)
required: false
example: "This is a voice notification from Home Assistant"
selector:
text:
multiline: true
duration:
name: Call Duration
description: How long to keep the call active (seconds)
required: false
default: 30
example: 45
selector:
number:
min: 5
max: 300
unit_of_measurement: seconds

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#!/usr/bin/env python3
"""SIP Voice Notifier Add-on Service."""
import json
import logging
import os
import tempfile
import time
from urllib.parse import urlparse
import requests
from flask import Flask, request, jsonify
from pydub import AudioSegment
import pjsua2 as pj
# Configure logging
logging.basicConfig(
level=logging.INFO,
format='%(asctime)s - %(name)s - %(levelname)s - %(message)s'
)
_LOGGER = logging.getLogger(__name__)
app = Flask(__name__)
# Global config
CONFIG = {}
DEFAULT_SAMPLE_RATE = 8000
class CallHandler(pj.Call):
"""Handle SIP call events."""
def __init__(self, account, call_id=pj.PJSUA_INVALID_ID):
pj.Call.__init__(self, account, call_id)
self.player = None
self.audio_file = None
self.connected = False
def onCallState(self, prm):
"""Called when call state changes."""
ci = self.getInfo()
_LOGGER.info(f"Call state: {ci.stateText}")
if ci.state == pj.PJSIP_INV_STATE_CONFIRMED:
_LOGGER.info("Call connected! Playing audio...")
self.connected = True
self.play_audio()
elif ci.state == pj.PJSIP_INV_STATE_DISCONNECTED:
_LOGGER.info("Call disconnected")
def onCallMediaState(self, prm):
"""Called when media state changes."""
ci = self.getInfo()
for mi in ci.media:
if mi.type == pj.PJMEDIA_TYPE_AUDIO and mi.status == pj.PJSUA_CALL_MEDIA_ACTIVE:
aud_med = self.getAudioMedia(mi.index)
try:
pj.Endpoint.instance().audDevManager().getPlaybackDevMedia().startTransmit(aud_med)
aud_med.startTransmit(pj.Endpoint.instance().audDevManager().getPlaybackDevMedia())
except Exception as e:
_LOGGER.warning(f"Audio routing warning: {e}")
def play_audio(self):
"""Play the audio file into the call."""
if not self.audio_file:
_LOGGER.error("No audio file specified")
return
try:
self.player = pj.AudioMediaPlayer()
self.player.createPlayer(self.audio_file, pj.PJMEDIA_FILE_NO_LOOP)
ci = self.getInfo()
for mi in ci.media:
if mi.type == pj.PJMEDIA_TYPE_AUDIO and mi.status == pj.PJSUA_CALL_MEDIA_ACTIVE:
aud_med = self.getAudioMedia(mi.index)
self.player.startTransmit(aud_med)
_LOGGER.info(f"Playing: {self.audio_file}")
break
except Exception as e:
_LOGGER.error(f"Error playing audio: {e}", exc_info=True)
def set_audio_file(self, audio_file: str):
"""Set the audio file to play."""
self.audio_file = audio_file
class SIPAccount(pj.Account):
"""SIP Account handler."""
def __init__(self):
pj.Account.__init__(self)
def onIncomingCall(self, prm):
"""Reject incoming calls."""
call = CallHandler(self, prm.callId)
call_param = pj.CallOpParam()
call_param.statusCode = 486
call.answer(call_param)
def download_and_convert_audio(url: str) -> str:
"""Download and convert audio to WAV."""
parsed_url = urlparse(url)
extension = os.path.splitext(parsed_url.path)[1] or ".mp3"
with tempfile.NamedTemporaryFile(delete=False, suffix=extension) as tmp:
download_path = tmp.name
response = requests.get(url, stream=True, timeout=30)
response.raise_for_status()
with open(download_path, 'wb') as f:
for chunk in response.iter_content(chunk_size=8192):
f.write(chunk)
_LOGGER.debug(f"Downloaded: {download_path}")
try:
audio = AudioSegment.from_file(download_path)
audio = audio.set_frame_rate(DEFAULT_SAMPLE_RATE)
audio = audio.set_channels(1)
audio = audio.set_sample_width(2)
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as tmp:
wav_path = tmp.name
audio.export(wav_path, format="wav")
_LOGGER.debug(f"Converted: {wav_path}")
os.remove(download_path)
return wav_path
except Exception as e:
if os.path.exists(download_path):
os.remove(download_path)
raise Exception(f"Audio conversion failed: {e}")
def generate_tts_audio(message: str) -> str:
"""Generate TTS audio."""
try:
from gtts import gTTS
with tempfile.NamedTemporaryFile(delete=False, suffix=".mp3") as tmp:
mp3_path = tmp.name
tts = gTTS(text=message, lang='en')
tts.save(mp3_path)
audio = AudioSegment.from_mp3(mp3_path)
audio = audio.set_frame_rate(DEFAULT_SAMPLE_RATE)
audio = audio.set_channels(1)
audio = audio.set_sample_width(2)
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as tmp:
wav_path = tmp.name
audio.export(wav_path, format="wav")
os.remove(mp3_path)
return wav_path
except ImportError:
raise Exception("gTTS not available")
def place_sip_call(destination: str, audio_file: str, duration: int):
"""Place a SIP call."""
ep = pj.Endpoint()
ep.libCreate()
ep_cfg = pj.EpConfig()
ep_cfg.logConfig.level = 3
ep_cfg.logConfig.consoleLevel = 0
ep.libInit(ep_cfg)
transport_cfg = pj.TransportConfig()
transport_cfg.port = 0
ep.transportCreate(pj.PJSIP_TRANSPORT_UDP, transport_cfg)
ep.libStart()
_LOGGER.info("PJSIP started")
try:
acc = SIPAccount()
acc_cfg = pj.AccountConfig()
sip_user = CONFIG['sip_user']
sip_server = CONFIG['sip_server']
sip_password = CONFIG['sip_password']
if not sip_user.startswith('sip:'):
sip_uri = f"sip:{sip_user}@{sip_server}"
else:
sip_uri = sip_user
acc_cfg.idUri = sip_uri
acc_cfg.regConfig.registrarUri = f"sip:{sip_server}"
cred = pj.AuthCredInfo("digest", "*", sip_user, 0, sip_password)
acc_cfg.sipConfig.authCreds.append(cred)
acc.create(acc_cfg)
_LOGGER.info(f"Account created: {sip_uri}")
time.sleep(2)
if not destination.startswith('sip:'):
dest_uri = f"sip:{destination}@{sip_server}"
else:
dest_uri = destination
_LOGGER.info(f"Calling: {dest_uri}")
call = CallHandler(acc)
call.set_audio_file(audio_file)
call_param = pj.CallOpParam()
call_param.opt.audioCount = 1
call_param.opt.videoCount = 0
call.makeCall(dest_uri, call_param)
wait_time = 0
while not call.connected and wait_time < 10:
time.sleep(0.5)
wait_time += 0.5
if not call.connected:
_LOGGER.warning("Call did not connect")
_LOGGER.info(f"Call active for {duration}s")
time.sleep(duration)
_LOGGER.info("Hanging up...")
hangup_param = pj.CallOpParam()
call.hangup(hangup_param)
time.sleep(1)
except Exception as e:
_LOGGER.error(f"Call error: {e}", exc_info=True)
raise
finally:
try:
ep.libDestroy()
except Exception as e:
_LOGGER.warning(f"Cleanup warning: {e}")
@app.route('/health', methods=['GET'])
def health():
"""Health check endpoint."""
return jsonify({'status': 'ok'}), 200
@app.route('/send_notification', methods=['POST'])
def handle_send_notification():
"""
Handle send_notification service call.
This endpoint is called by Home Assistant when the service is invoked.
"""
try:
data = request.json
destination = data.get('destination')
audio_url = data.get('audio_url')
message = data.get('message')
duration = data.get('duration', CONFIG.get('default_duration', 30))
if not destination:
return jsonify({'error': 'destination required'}), 400
if not audio_url and not message:
return jsonify({'error': 'audio_url or message required'}), 400
temp_files = []
try:
# Prepare audio
if audio_url:
_LOGGER.info(f"Downloading: {audio_url}")
audio_file = download_and_convert_audio(audio_url)
temp_files.append(audio_file)
elif message:
_LOGGER.info(f"Generating TTS: {message}")
audio_file = generate_tts_audio(message)
temp_files.append(audio_file)
# Place call
_LOGGER.info(f"Calling: {destination}")
place_sip_call(destination, audio_file, duration)
return jsonify({'status': 'success'}), 200
finally:
# Cleanup
for f in temp_files:
try:
if os.path.exists(f):
os.remove(f)
except Exception as e:
_LOGGER.warning(f"Cleanup failed: {e}")
except Exception as e:
_LOGGER.error(f"Request error: {e}", exc_info=True)
return jsonify({'error': str(e)}), 500
if __name__ == '__main__':
# Load config from Home Assistant add-on options
options_file = '/data/options.json'
if os.path.exists(options_file):
with open(options_file, 'r') as f:
CONFIG = json.load(f)
_LOGGER.info("Config loaded from options.json")
else:
_LOGGER.warning("No options.json found, using environment variables")
CONFIG = {
'sip_server': os.getenv('SIP_SERVER', ''),
'sip_user': os.getenv('SIP_USER', ''),
'sip_password': os.getenv('SIP_PASSWORD', ''),
'default_duration': int(os.getenv('DEFAULT_DURATION', 30))
}
_LOGGER.info("SIP Voice Notifier ready - service will be auto-registered by Supervisor")
_LOGGER.info("Starting service on port 8099")
app.run(host='0.0.0.0', port=8099, debug=False)