AIPBX：开源企业级 IP PBX 管理门户与统一通信平台

AI PBX

Open-Source Enterprise IP PBX Management Portal & Unified Communications
Asterisk 22 · PHP 8 · MariaDB · WebRTC · Instant Messaging · Android & iOS Apps

📖 Documentation · 🌐 Website: aipbx.bid

Quick Install • Updating • Features • Documentation • Architecture • Changelog • Roadmap • Contributing

Quick Install

# One-liner installation (installs the latest release)
curl -fsSL https://raw.githubusercontent.com/mahirgul/AiPBX/main/install.sh | sudo bash

Or clone manually and install the latest release:

git clone https://github.com/mahirgul/AiPBX.git /opt/aipbx
cd /opt/aipbx
git checkout "$(git tag -l 'v*' --sort=-v:refname | head -1)"
sudo bash install.sh

The installer:

1. Asks for your FQDN — with Let's Encrypt if DNS points to the server, otherwise a self-signed certificate; leave it blank for a LAN-only install announced as aipbx.local via mDNS
2. Generates strong random secrets for every service (no default passwords)
3. Installs and configures Asterisk, MariaDB, Nginx + Apache, PHP, coturn, the Go chat service, firewalld and fail2ban, and builds the database from migrations
4. Prints the credentials and saves them to /root/aipbx-credentials.txt

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

Developers: AIPBX_REF=main installs the current development branch instead of the latest release.

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

Updating

Releases are published as git tags (vX.Y.Z) with notes in CHANGELOG.md.

sudo aipbx-update --check   # installed vs. latest release
sudo aipbx-update           # update to the latest release

Or from the portal: Admin → System Update (admin only) shows the installed and latest version, the release notes and an Update button with live progress. A new release is also checked for every night.

Every update:

• refuses to start during active calls (Asterisk is restarted) or over hand-edited code;
• backs up the database, /etc/ai-pbx.env and /etc/asterisk to /var/backups/aipbx/;
• applies database migrations, new packages and system settings without touching passwords, certificates, the admin password or your firewall choices;
• verifies services, the portal and every page — and rolls back automatically (code, database and settings) if anything fails.

Keep customisations out of /opt/aipbx (use the portal, /etc/ai-pbx.env and *_custom.conf files) so they survive updates. See INSTALL.md for details.

Features

📞 PBX Management

• Extension management — PJSIP-based, one number on desk phone, browser and mobile at once (SIP + WebRTC + mobile endpoints)
• Calling permission groups (call barring) — prefix and exact-match rules (0, 05, 00), per-extension assignment, default allow-all
• Voicemail — Asterisk voicemail with unconditional / busy / no-answer / unavailable triggers, managed in My Phone or with feature codes (*97 own mailbox, *98 mailbox login), in-browser player
• Boss – secretary groups — executive interception, simultaneous or sequential secretary ringing, VIP whitelist bypass, automatic failover
• Ring groups — virtual numbers mixing extensions and external numbers, simultaneous or sequential
• Conference rooms — ConfBridge with PINs, wait-for-leader, join muted and live web moderation (mute/kick)
• Trunks — IP or registration based, tabbed settings, copy, rename and drag-and-drop ordering; inbound DID trimming, transit routing between PBXes and carriers, per-trunk outbound caller ID normalization (keep last N digits + prefix, e.g. extension 7840 → 903704187840)
• Outbound routes — Asterisk patterns, strip/prepend, trunk failover chain, route groups for PBXes with different number formats, copy and ordering, duplicate patterns rejected
• DID routing, time conditions, multi-level IVR
• Queues — dynamic agent login/logout, static agents, hold music, recording
• Feature codes — *81/*80 queue login/logout, *72 call forward, *60 DND, *43 intercom, *90/*91/*92 spy/whisper/barge, *97/*98 voicemail
• In-band disconnect supervision — disconnect-tone detection for analog/legacy trunks
• Safe apply — generated configuration is reloaded atomically; a failed write or Asterisk reload is rolled back and the change stays pending
• Live dashboard — active calls and channels, callers waiting in queues, today's answered / missed calls and per-trunk channel usage, refreshed every 5 seconds
• Complete Turkish prompt set in one voice — every Asterisk core prompt plus voicemail, digits, conference, queue and directory prompts, so Turkish calls never switch to English halfway. Shipped as asterisk-core-sounds-tr packages in six formats (wav, ulaw, alaw, gsm and HD g722/sln16), so Asterisk plays the file matching the call's codec — and any Asterisk server can use them
• Call recordings — converted automatically to mono 16 kbps MP3 (~8× smaller than WAV), played in CDR reports, My Phone and the apps

📠 Fax

• Inbound/outbound fax — T.38 and G.711 (res_fax + SpanDSP)
• Compose in the browser (rich-text editor) or upload a PDF; resend failed faxes
• E-mail notification with the fax attached; per-DID recipients

📊 Call Center & Supervision

• Agent desk and wallboard — live queues, waiting callers (with pickup), agent states (idle / ringing / in call / on break), SLA, answered / missed / abandoned statistics
• Breaks with reasons, applied in Asterisk per queue, and break reports
• Listen, whisper and barge — from the wallboard or with *90/*91/*92
• Call transfer that keeps the caller connected through queue Local channels
• Queue Report Centre — service level, answered / lost calls, average wait and talk time per queue and per agent, lost calls with call-back tracking, repeat callers and call outcomes
• Call reports and call journey — calls grouped by linkedid with a step-by-step timeline; caller, inbound and outbound trunk, dialled number and direction (inbound / outbound / internal / trunk to trunk, transferred), recording playback with a waveform player
• Branded PDF and Excel export for every report, as filtered, with all rows

🌐 WebRTC Softphone

• Browser phone built into the portal header — no installation
• TURNS (coturn) relay for clients behind NAT and strict firewalls — clients use 443, which the nginx multiplexer hands to coturn (5349); per-user, time-limited credentials refreshed automatically
• NAT handled in the generator — public/private address mapping for ICE and every SIP transport; see WebRTC, NAT and TURN
• Opus + DTLS-SRTP encrypted audio

💬 Messaging & Group Chat

• 1-to-1 and group chat (Go WebSocket service aipbx-chat), group admins, member management
• Photos and files with thumbnails; access limited to the conversation's participants
• Presence, typing indicators, read receipts; push notifications on mobile

📱 Mobile Apps

• Android (Kotlin) — WebRTC calling with push wake-up and a foreground service, chat, directory with presence, call history, DND / call forward, in-app log viewer
• iOS (SwiftUI, CallKit) — calling with native incoming-call screen, chat, directory, features
• Sign-in with username + password (+ 2FA code), by scanning a QR code on the portal, or with the one-time link in the invitation e-mail

🤖 AI

• Cloud TTS — turn text into speech with Google (API key or service account), Amazon Polly, Azure AI Speech, ElevenLabs or OpenAI; listen, download the MP3 or save it as an announcement in one click. Credentials are stored encrypted; usage is tracked per character

💼 Microsoft Teams Integration

• Direct Routing over SIP-TLS 5061 with SRTP
• User ↔ extension mapping (Microsoft 365 UPN, E.164)
• Teams channel notifications (missed calls, voicemail, queue alarms, faxes)
• Generated PowerShell setup scripts from your PBX configuration

🔐 Authentication & Security

• Two-factor authentication (TOTP) with offline QR codes and single-use recovery codes — enforced for portal, mobile and Google sign-in
• Passkeys (WebAuthn / FIDO2) — passwordless sign-in with biometrics or PIN
• Google sign-in (optional) with token audience and verified e-mail checks
• Sessions revoked on password change — mobile and chat tokens stop working immediately
• User invitations — generated passwords, forced first-login change, mobile sign-in link
• RBAC — built-in roles (admin, viewer, queue manager, agent, fax, user — the default) and custom roles with per-module permissions; security-critical pages are admin-only
• Brute-force protection — math CAPTCHA, account/IP lockout and a fail2ban jail using the real client address (PROXY protocol from the 443 edge)
• Certificates page — one certificate for portal, TURNS and SIP-TLS: Let's Encrypt (tested on staging first, renewed automatically), upload PEM or PFX/P12, or self-signed; problems that silently break WebRTC (self-signed, wrong name, missing chain) are flagged
• Nightly backups — database and configuration (aipbx-backup, 14 days, optional off-server copy)
• Firewall and fail2ban management from the portal through a single, argument-validated root helper (aipbx-priv) — the portal never runs arbitrary commands as root
• Hardening — CSRF tokens, security headers, sandboxed uploaded files, Asterisk HTTP on loopback, Apache kept in Ubuntu's systemd sandbox with only the paths the portal needs opened

🌍 Multi-language

• Turkish 🇹🇷 and English 🇬🇧 (2,000+ keys, parity checked in CI)

Architecture

Internet ──443──▶ nginx (stream, ALPN, PROXY protocol)
                    ├─ HTTPS / WSS ──▶ Apache + PHP (127.0.0.1:8443)
                    │                    ├─ /ws       ──▶ Asterisk WebSocket (127.0.0.1:8088)
                    │                    └─ /chat/…   ──▶ aipbx-chat (127.0.0.1:8086)
                    └─ TURNS ─────────▶ coturn
         ──5060/5061, RTP──▶ Asterisk 22 (PJSIP)  ──ODBC──▶ MariaDB (CDR, queue_log)

AiPBX/
├── web/                    # PHP MVC portal
│   ├── src/
│   │   ├── controllers/    # 53 page controllers
│   │   ├── services/       # 45 services (business logic)
│   │   ├── repositories/   # 33 repositories
│   │   └── sync/           # 19 Asterisk configuration generators
│   ├── templates/          # layouts + 51 views
│   ├── api/                # JSON endpoints (call control, mobile, chat token, WebAuthn…)
│   ├── lang/               # tr / en
│   ├── db/migrations/      # Phinx migrations (the database schema)
│   └── tests/              # PHPUnit
├── chat/                   # Go WebSocket chat service
├── android/                # Kotlin app
├── ios/                    # SwiftUI app
├── asterisk-config/        # Asterisk base configuration
├── sounds/                 # bundled custom sounds + Turkish prompt texts (core-sounds-tr.txt)
├── asterisk-core-sounds-tr-*.tar.xz  # Turkish prompts, one package per format (+ SHA256SUMS)
├── docs/                   # administrator documentation
├── conf/sbin/              # aipbx-priv (root helper), aipbx-update (updater)
├── db/seed.sql             # initial roles, permissions, settings
├── scripts/release.sh      # publishes a release
├── install.sh              # installer (--upgrade mode used by aipbx-update)
├── VERSION · CHANGELOG.md
└── README.md · INSTALL.md · ARCHITECTURE.md

Technology Stack

After Installing

• Credentials are in /root/aipbx-credentials.txt — note them and delete the file.
• Service secrets live in /etc/ai-pbx.env.
• To switch a self-signed install to a Let's Encrypt or your own certificate, use Security → Certificates — see docs/certificates.md.
• Ports and updating: INSTALL.md.
• Set an e-mail relay (fax-to-e-mail, voicemail, invitations): docs/mail.md.
• Configuring trunks, routes, roles and more: documentation.

Contributing

Contributions are welcome — see CONTRIBUTING.md for code style, commit conventions and testing.

1. Fork the repository
2. Create a feature branch (git checkout -b feat/my-feature)
3. Commit your changes (git commit -m 'feat: add awesome feature')
4. Push the branch and open a Pull Request

Code comments, commit messages and documentation are written in English.

License

MIT License.

Author & Creator

Mahir Gül · mhrgl.com · @mahirgul