Modern Open-Source Web GUI for Asterisk PBX
Manage your phone system through an intuitive web interface – extensions, SIP trunks, call routing, voicemail, and real-time monitoring.
Website • Live Demo • Quick Start • Screenshots • Features • Changelog • Support the Project
- 📞 Extension Management – Create, edit, and manage SIP extensions with caller ID, context, and activation status
- 📖 Phonebook – Global and per-extension address books with CSV import/export
- 🔌 SIP Trunk Configuration – Connect to any SIP provider via registration or IP authentication, with built-in templates for Plusnet IPfonie, Telekom DeutschlandLAN, CompanyFlex, Telekom All-IP (Privatkundenanschluss) and Iliad (Italy)
- 📠 DID Routing – Flexibly assign incoming phone numbers to extensions with number block management per trunk
- 📤 Outbound CID Selection – Choose which assigned DID to use as outbound caller-ID per extension via dropdown
- 🆔 P-Asserted-Identity (PAI) – Optional PAI header per extension (e.g. main number of a number block)
- 🔄 Call Forwarding – Unconditional, busy, and no-answer forwarding per extension, toggled with one click
- 🔔 BLF & Pickup Groups – Busy lamp field hints and call pickup groups per extension
- 📞 Ring Groups (Sammelruf) – Create ring groups with strategies and assign inbound numbers
- 🎛️ IVR – Multi-level IVR menus with prompt upload/conversion, retries, and inbound DID assignment
- 📩 Voicemail – Per-extension voicemail boxes with PIN, email notifications (HTML), configurable ring timeout, and built-in audio player
- 📧 SMTP Email Configuration – Configure your mail server for voicemail-to-email delivery, with built-in test email function (Port 465/587 auto-detection)
- 🏠 Home Assistant Integration – API-key authentication, MQTT publisher for call events, and click-to-call via originate endpoint
- 🎵 Music on Hold – Pre-installed hold music in high-quality g722 format
- 🎙️ German Voice Prompts – Built-in German Asterisk sound pack for IVR and voicemail announcements
- 🔒 IP Whitelist – Restrict SIP registration to trusted IP addresses and CIDR networks
- 🔑 SIP Password Strength – Real-time password strength indicator, secure password generator, and weak password overview
- 📝 Audit Log – Full audit trail of all administrative actions with color-coded log viewer
- 🛡️ Fail2Ban Integration – Live Fail2Ban status with active bans, jail overview, and recent ban history
- 🎛️ Codec Management – Global and per-extension codec configuration with drag-and-drop priority ordering
- 📊 Call Detail Records – Full CDR with filters by source, destination, and status, plus call statistics at a glance
- 🔐 Multi-User & Roles – Admin and user roles with JWT-based authentication
- 📡 Real-Time Dashboard – Live overview via WebSocket: Asterisk status, registered endpoints, active lines, and recent calls
- 🌙 Dark Mode – System-wide dark theme with OS preference detection, manual toggle, and localStorage persistence
- 🔍 SIP Debug – Live SIP message viewer (INVITE, BYE, 200 OK) with per-call grouping, expandable raw SIP text, and toggle capture (admin-only)
- 🌐 Multilingual UI (DE/EN) – Choose German or English during installation; UI is fully translated
- 🔄 One-Click Updates – Update GonoPBX directly from the web GUI or via command line
- 🐳 Docker Deployment – Full system up and running in minutes with
docker compose up
| Dashboard | Settings | Security |
|---|---|---|
|
 |
 |
| Email Settings | Audio Codecs | Server Management |
|---|---|---|
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- Linux server (Ubuntu 22.04+ / Debian 12+ recommended)
- Docker & Docker Compose installed
- Ports 3000 (Web UI), 5060 (SIP), 10000-20000 (RTP) available
# 1. Clone the repository
git clone https://github.com/ankaios76/gonopbx.git
cd gonopbx
# 2. Run the interactive installer
chmod +x install.sh
./install.sh
# 3. Access the web interface
# Open https://your-server-ip:3000 in your browserThe installer will automatically:
- Detect your server IP
- Generate secure passwords
- Create the Docker configuration
- Ask for the UI language (German or English)
- Start all services
| Component | Technology |
|---|---|
| PBX Engine | Asterisk 20 (PJSIP) |
| Backend | FastAPI (Python) |
| Frontend | React + TypeScript (Vite, Tailwind CSS) |
| Database | PostgreSQL |
| Auth | JWT + bcrypt |
| Real-Time | WebSocket |
| msmtp (in Asterisk container) | |
| Deployment | Docker Compose |
gonopbx/
├── asterisk/ # Asterisk Dockerfile + configuration templates
├── backend/ # FastAPI backend (API, WebSocket, Asterisk integration)
├── frontend/ # React frontend (Vite + Tailwind)
├── docker-compose.yml # Container orchestration
├── install.sh # Interactive installer
└── CHANGELOG.md # Version history
Go to Settings → Server → Update and click "Update installieren". The system will pull the latest version, rebuild containers, and restart automatically.
cd /root/asterisk-pbx-gui
git pull origin main
docker compose up -d --buildDatabase migrations run automatically on startup — no manual steps required.
New Features:
- Multilingual UI (DE/EN) – UI translation framework with installer language selection
New Features:
- Iliad (Italy) SIP Trunk – New provider profile with preset registrar/proxy
voip.iliad.itand From-User (login number) support
New Features:
- Telekom All-IP (Privatkundenanschluss) – New provider profile for Telekom residential connections (MagentaZuhause). Supports
P-Preferred-Identityheader, TCP transport, andtel.t-online.deregistrar. Separate fields for Zugangsnummer (auth) and Anschlussnummer (From-User, E.164 format)
New Features:
- Phonebook – Global and per-extension address books with CSV import/export
- BLF & Pickup Groups – BLF hints and pickup groups per extension
- Ring Groups (Sammelruf) – Ring groups with strategies and inbound DID assignment
- IVR Enhancements – Prompt upload with conversion/validation, retries, and inbound DID mapping
- UI Improvements – SIP Debug moved to Settings, Audio tab renamed to Audio-Codecs
New Features:
- SIP Debug – Live SIP message viewer in the web GUI. Toggle capture on/off, messages grouped by Call-ID, expandable raw SIP text with direction arrows and color-coded methods/status codes. Based on Asterisk's
res_pjsip_historymodule with 3-second polling. Messages kept for 2 hours (max 10,000). Admin-only
New Features:
- Outbound CID Selection – Choose which assigned DID to use as outbound caller-ID per extension (dropdown in extension detail)
- P-Asserted-Identity (PAI) – Optional PAI header per extension, e.g. main number of a number block
- Home Assistant Integration – API-key auth, MQTT publisher for call events, click-to-call via originate endpoint
- Home Assistant Settings – Configure MQTT broker, API key, and test connection from the web GUI
- Website redesign with updated screenshots
New Features:
- Dark Mode – Full dark mode with automatic OS preference detection and manual toggle (Sun/Moon icon in header). Persists via localStorage
- Provider logos in trunk cards now displayed as round avatars with white background
- "Made with ❤️ in Bremen" added to footer
Improvements:
- Dashboard redesign: personalized greeting (Moin/Hallo/Guten Abend) with username, weekday and date
- Version display moved from footer to Dashboard status card
- dus.net SIP server corrected to proxy.dus.net
- System Health card removed from Dashboard
New Features:
- User profiles with full name, avatar upload and extension assignment
- Multi-DID routing: multiple inbound numbers per extension
- Welcome email with credentials for new users
- Self-service password change for all users
- FAQ page built into the web interface
- Navigation restructured: Extensions and Trunks moved under Settings
- One-click system update from the web GUI
Bugfixes:
- Fixed user creation error (removed unique email constraint)
New Features:
- SIP password strength checker with real-time indicator (red/yellow/green bar)
- Secure password generator (16 characters) with one-click button
- Weak password overview in Security settings tab
- Full audit log for all administrative actions (create, update, delete)
- Audit Log tab in Settings with color-coded entries and pagination
- Fail2Ban live status in Security tab (active bans, jails, recent bans)
New Features:
- IP whitelist for SIP registration (restrict to trusted IPs/CIDRs)
- Music on Hold with pre-installed g722 audio files
- HTML voicemail email notifications
- Per-extension ring timeout (configurable seconds before voicemail picks up)
- German voice prompts (asterisk-prompt-de)
Improvements:
- Auto-detection of external IP address
- SMTP port 465 implicit TLS support
- Dialplan: device-state check before dial, early answer for inbound trunk calls
- DID extraction from SIP To-header as fallback
- Duplicate identify section protection for multi-trunk setups
- Installer: Docker permission check
- Global and per-extension codec management with drag-and-drop priority
- SMTP email configuration with test email function
- Per-extension voicemail mailbox management
- Custom SIP trunk provider support
Initial release with full PBX management: extensions, trunks, routing, call forwarding, voicemail, CDR, authentication, and real-time dashboard.
See CHANGELOG.md for the full version history.
Contributions are welcome! Whether it's bug reports, feature requests, or pull requests – all help is appreciated.
- Fork the repository
- Create your feature branch (
git checkout -b feature/amazing-feature) - Commit your changes (
git commit -m 'Add amazing feature') - Push to the branch (
git push origin feature/amazing-feature) - Open a Pull Request
Have an idea but no time to code? Open an issue – I'll implement it when I find the time.
- Multi-database support (SQLite for home use, MySQL/MariaDB)
- Ring groups & call queues
- IVR / auto attendant builder
- Conference rooms
- Phonebook with CallerID lookup
- Multi-language support (EN/DE)
- Backup & restore functionality
This project is open source and available under the MIT License.
GonoPBX is free and open source. If you find it useful, please consider:
- ⭐ Starring this repository – it helps with visibility
- 🐛 Reporting bugs or suggesting features via Issues
- ☕ Buy me a Coffee – helps cover hosting costs
Made with ❤️ by Norbert Hengsteler
gonopbx.de