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FreePBX / Asterisk

FreePBX with Asterisk is an open-source phone system (PBX). With it, you can operate VoIP phones, automate calls, and build a professional telephony infrastructure.

For Advanced Users

FreePBX requires VoIP knowledge. You need a SIP provider or VoIP phones for meaningful use.

Installation

Add the following template to your docker-compose.yml and then run ei23 dc.

Template

  freepbx-app:
    image: epandi/asterisk-freepbx-arm:17.15-latest
    container_name: freepbx-app
    restart: unless-stopped
    ports:
      - 2233:80           # Web interface
      - 5060:5060/udp     # SIP
      - 5160:5160/udp     # SIP Alt
      - 18000-18100:18000-18100/udp  # RTP (call data)
      - 4445:4445         # Flash Operator Panel
    volumes:
      - ./volumes/asterisk17/certs:/certs
      - ./volumes/asterisk17/data:/data
      - ./volumes/asterisk17/logs:/var/log
      - ./volumes/asterisk17/data/www:/var/www/html
      - ./volumes/asterisk17/db:/var/lib/mysql
    environment:
      - VIRTUAL_HOST=asterisk.local
      - VIRTUAL_PORT=80
      - ZABBIX_HOSTNAME=freepbx-app
      - RTP_START=18000
      - RTP_FINISH=18100
      - DB_EMBEDDED=TRUE
    cap_add:
      - NET_ADMIN

First Start

  1. After starting, access FreePBX at http://[IP]:2233
  2. Follow the setup wizard
  3. Create an admin password

Basic Configuration

SIP Trunk (SIP Provider)

  1. Go to Connectivity → Trunks
  2. Click Add Trunk → Add SIP (chan_pjsip) Trunk
  3. Configure:
    • Trunk Name: Provider name
    • Outbound CallerID: Your phone number
    • SIP Settings: Server, user, password from provider

Extensions

  1. Applications → Extensions
  2. Add Extension → Add New Chan_PJSIP Extension
  3. Configure:
    • Extension Number: e.g., 1001
    • Display Name: Name
    • Secret: Password for the phone

Inbound Routes

  1. Connectivity → Inbound Routes
  2. Define what happens on incoming calls
  3. Destination: Extension, IVR, Voicemail, etc.

Outbound Routes

  1. Connectivity → Outbound Routes
  2. Configure which extensions use which trunks

Use Cases

Application Description
Home Office Own phone system for home office
Door Intercom Connect SIP intercom
Emergency Phones Old landline phones via SIP
Voicemail Set up answering machine
IVR Voice menu "Press 1 for..."

Notes

  • FreePBX accessible on port 2233
  • SIP ports 5060/5160 may need port forwarding on router
  • RTP ports 18000-18100 for voice data
  • Data in ./volumes/asterisk17/
  • Image optimized for ARM (Raspberry Pi)

Alternatives

For simple SIP use, a simple SIP client is often sufficient. FreePBX is worthwhile with multiple phones or complex call routes.

Further Information