Open-Source Enterprise IP PBX Management Portal & Unified Communications

Asterisk 22 · PHP 8 · MariaDB · WebRTC · Instant Messaging · Android & iOS Apps

Quick Install • Features • Architecture • Website • Contributing

# Clone the repository

git clone https://github.com/mahirgul/AiPBX.git /opt/aipbx

cd /opt/aipbx

# Run the installer as root

sudo bash install.shThe installer will:

- Ask for your FQDN (domain name) — or use your IP with a self-signed cert

- Generate strong random passwords for all services automatically

- Install and configure everything (Asterisk, MariaDB, Apache2, coturn, Chat service)

- Display all credentials at the end and save them to /root/aipbx-credentials.txt

Requirements: Ubuntu 22.04 / 24.04 / 26.04 LTS · 2 GB RAM · 10 GB disk · root access

After install, open https://<your-server> in your browser and log in with the credentials shown.

- Extension management — PJSIP-based, dual-endpoint (SIP + WebRTC)

- Trunk management — dynamic PJSIP trunk configuration

- Call routing — DID mapping, outbound routes, time conditions

- IVR — multi-level voice menus with time-based routing

- Queue management — call queues, dynamic agent login/logout, hold music

- Feature codes — *81queue login (all or*81<queue>),*80queue logout,*72call forward,*60DND,*43intercom,*88spy/whisper

- In-band disconnect supervision — cadence-based disconnect tone detection (from-trunk-kapanma-tonu) for analog/legacy trunks without out-of-band hangup signaling

- Auto-rollback — failed Asterisk reloads are automatically reverted

- Inbound/outbound fax — T.38 and G.711 (res_fax + SpanDSP)

- WYSIWYG fax editor — compose rich-text faxes directly in the browser

- PDF upload & send

- Fax retry

- Email notification — incoming faxes forwarded by email automatically

- Real-time agent panel — live queue status, active calls, agent break selector

- Dynamic queue login/logout — via star codes (*81/*80) with audio confirmation (queue-agentlogin-success/ beeps) or web UI

- Intelligent call transfer — bridge-traversal caller preservation (findCallerChannelForAgent) prevents dropped lines during attended transfer

- Auto-desk navigation — answering incoming queue calls in WebRTC automatically shifts SPA view to agent CRM/notes (/cc-agent)

- CDR reporting — detailed call records, filtering, export

- Call recording playback — listen to recordings in the browser

- In-browser SIP phone — zero-install browser phone embedded directly in topbar

- Resilient TURN/STUN — 30-minute automatic credential renewal eliminates silent audio on extended shifts

- Opus + DTLS-SRTP — high-quality, end-to-end encrypted audio

- 1-to-1 & Multi-user Group Rooms — Real-time team messaging powered by high-performance Go WebSocket daemon (aipbx-chat)

- Rich Group Management — Create groups, assign admin roles, invite/remove participants (up to 256 members per group)

- Media & File Sharing — Image compression, thumbnail generation, and secure attachment delivery

- System Audit Trail — System-generated messages for member additions, removals, and role updates

- Presence & Delivery Receipts — Real-time typing indicators, read receipts, and online status tracking

- FCM Push Notifications — Background notifications with conversation grouping for both direct and group messages

- Native Kotlin application with zero external cloud dependencies

- PJSIP + WebRTC dual engine with Opus HD audio & DTLS-SRTP encryption

- FCM Push & Persistent Foreground Service — instantaneous wake-up for incoming calls

- Full Group Chat & Instant Messaging — Direct 1-to-1 messaging, multi-user group chat rooms, quick group creation (+ Yeni Grup), conversation filter chips (All, Direct, Groups), participant management, admin roles, and real-time WebSocket updates

- Corporate Directory & Live Presence — 50+ extensions with live status & 1-tap dialing

- Detailed Call History & In-App Log Viewer — full diagnostics and call filtering

- Available on Google Play Console — automated Closed Testing track pipeline and direct signed APK downloads

- Native SwiftUI modern application for iPhone & iPad (iOS 16.0+)

- Embedded WebRTC Voice Engine with zero external cloud dependencies

- CallKit & AudioSession Integration — Native iOS lock-screen incoming calls, audio routing, speaker and mute

- Full Group Chat & Direct Messaging — Instant messaging, multi-party group rooms, contact picker, and real-time WebSocket communication

- Corporate Directory & Live Presence — 50+ extensions with live presence indicators and 1-tap call/chat

- PBX Features & In-App Diagnostics — DND, Call Forwarding, live log viewer, and log sharing via iOS ShareSheet

- Automated GitHub Actions CI/CD — compiled automatically on macOS runners into unsigned .ipaand Simulator.zipartifacts

- WhatsApp/Telegram-Style Master-Detail Chat — Fluid responsive navigation on smartphones (<= 768px) with hardware/browser back-button popstate support

- Optimized Mobile Views — Touch-friendly responsive layouts for My Phone (/my_phone), Role Permission Matrix (/roles), Mobile Push Settings (/push-settings), and Pending Sync (/pending-sync)

- High-concurrency Nginx support — event-driven WebSocket multiplexer, TLS 1.3, PHP-FPM FastCGI

- Native Apache 2.4 support — mod_proxy_wstunneland.htaccesscompatibility

- Unified Port 443 — multiplexes Asterisk WebRTC SIP (/ws) and Go Chat (/chat/ws)

- Ready-to-use Nginx template — available in conf/nginx/aipbx.conf.example

- Granular RBAC — modular role-permission matrix (sys_role_permissions) with strict read-only viewer mode, includingmy_phoneandchatcontrols

- Math CAPTCHA + brute-force lockout (5 failures → 15-min IP ban)

- CSRF protection — token on every POST form

- fail2ban integration

- Firewall management — control firewalld/fail2ban from the web UI

- Credentials served via API — never embedded in page source

- Turkish 🇹🇷 and English 🇬🇧 (1,300+ translation keys)

- Easy to extend with the t()function

AiPBX/

├── conf/ # Nginx & web server production templates

│ └── nginx/aipbx.conf.example

│

├── web/ # PHP MVC Web Portal

│ ├── src/

│ │ ├── controllers/ # 35 page controllers

│ │ ├── services/ # 24 business logic services

│ │ ├── repositories/ # 28 database repositories

│ │ └── sync/ # 13 Asterisk config generators

│ ├── templates/views/ # 34 PHP view templates

│ ├── api/ # REST API layer

│ ├── assets/ # CSS, JS, fonts

│ ├── lang/ # Language files (tr/en)

│ └── db/migrations/ # Phinx database migrations

│

├── android/ # Kotlin Android App

│ └── app/src/main/

│ └── java/com/mhrgl/aipbx/

│

├── ios/ # Swift & SwiftUI iOS App (GitHub Actions CI/CD)

│ ├── AiPBX/ # App, Models, Services, Views, Resources

│ └── AiPBX.xcodeproj/ # Xcode project & schemes

│

├── chat/ # Go WebSocket Chat Service

│ ├── main.go

│ ├── hub.go # WebSocket hub

│ ├── handlers.go # HTTP/WS handlers

│ └── db.go # Database layer

│

├── asterisk-config/ # Asterisk reference configuration

│ └── pbx/ # Modular dialplan, PJSIP, queue files

│

├── db/ # Database schema

│ ├── schema.sql # Table definitions

│ └── seed.sql # Initial seed data

│

├── install.sh # One-command installer

└── README.md

All credentials (admin password, DB passwords, AMI key, TURN secret) are generated randomly during install and saved to /root/aipbx-credentials.txt.

You can change all passwords later from Admin Panel → Settings → System.

To add a real TLS certificate after install (if you skipped Let's Encrypt):

certbot --apache -d your-domain.comContributions are welcome! Please check out our Contributing Guidelines for details on code style, commit conventions, and sandbox testing.

- Fork the repository

- Create a feature branch (git checkout -b feat/my-feature)

- Commit your changes (git commit -m 'feat: add awesome feature')

- Push to the branch (git push origin feat/my-feature)

- Open a Pull Request

This project is licensed under the MIT License.