callytics

Self-hosted open source call center platform. One command installs everything on Linux via Docker.

What it is

callytics is a self-hosted call center platform for developers and small businesses who need programmable IVR, call routing, SIP trunks, and a live operations dashboard without Twilio pricing or FreePBX complexity.

It is built around Asterisk ARI + Stasis, so call flows are database-driven and update instantly from the UI, with no manual dialplan editing required.

Features

• Visual IVR flow builder with 13 node types
• SIP extensions, trunks, and inbound DID routing
• Outbound call campaigns with CSV upload and sliding window dialer
• Call queues and operator management
• Hunt groups with sequential, random, and group dial strategies
• Conference rooms via Asterisk ConfBridge
• Callback node for caller-requested callbacks
• Audio upload, ffmpeg conversion, and offline Piper TTS
• Call recordings with browser preview and download
• Live dashboard with active calls, queue status, and recent events
• Call logs with execution trace and RTP quality scoring
• SIP capture with live packet stream, ladder diagrams, and pcap export
• Network diagnostics with trunk testing, SIP registration status, and resource usage
• Asterisk log viewer with plain-English translation
• SIP firewall with auto-blocking, live feed, and GeoIP
• Backup and restore with scheduling and retention
• WireGuard VPN with peer management, QR onboarding, and VPN-only extension controls
• IVR templates with one-click import and JSON import
• Network preflight wizard

Quick start

git clone https://github.com/rayanweragala/callytics.git
cd callytics
cp .env.example .env
bash scripts/install.sh

Required configuration before first start

Before running bash scripts/install.sh, copy .env.example to .env and set these values:

• HOST_IP - set to your machine's LAN IP address, not 127.0.0.1. Run ip route get 1 | awk '{print $7; exit}' to find it. Leaving it as 127.0.0.1 means softphones on other devices will fail to register and SIP QR codes will be wrong.
• RECORDINGS_INTERNAL_TOKEN - change from the default value. Used to authenticate internal recording requests between Stasis and backend.
• SIP_PORT - default 5080. Only change if that port is already in use on your host.

Then open http://localhost:3000

Requirements

• Linux only (Ubuntu 22.04+ recommended)
• Docker and Docker Compose
• Ports: 3000 (UI), 3001 (API), 5080 (SIP), 8088 (ARI), 51820/udp (WireGuard optional)

Supported environments

• Ubuntu 22.04 and 24.04 tested and supported
• Linux only
• Docker 24+ required

Tech stack

• Asterisk 20
• Node.js
• NestJS
• React + Vite
• PostgreSQL
• Redis
• Docker Compose

License

MIT

Credits

• Asterisk — GPLv2 telephony engine used in the runtime stack.
• NGINX — BSD 2-Clause web server used in the frontend runtime image.
• Sniffnet — design inspiration for the UI theme and color system.

Troubleshooting

• Port already in use: check which process owns the port with ss -tulpn | grep <port>, stop it, then rerun docker compose up -d.
• SIP not registering: verify SIP_PORT is 5080 and not blocked by firewall, confirm extension credentials match what is provisioned in the app.
• Audio not playing: check that the audio file was uploaded and converted successfully in the Audio page, verify Asterisk container is healthy.
• One-way audio: disable SIP ALG on your router, check NAT detection result in the preflight wizard.
• Docker socket permission denied: ensure the user running Docker is in the docker group (sudo usermod -aG docker $USER).