How to Run a VoIP Server on a VPS with FreePBX and Asterisk

Running your own VoIP (Voice over IP) phone system on a VPS gives small businesses and teams a professional private branch exchange (PBX) at a fraction of the cost of hosted PBX services. Asterisk is the world’s most widely used open-source PBX engine, and FreePBX provides the browser-based management interface on top of it — together they power millions of business phone systems worldwide.

This guide installs FreePBX with Asterisk on an Ubuntu VPS, configures SIP extensions, sets up a SIP trunk for real phone calls, adds voicemail, and hardens the installation against VoIP fraud.

What You Can Build

• Internal extensions for your team (100, 101, 102…)
• Real inbound/outbound phone numbers via SIP trunk
• Voicemail with email delivery
• Call queues and IVR menus (“Press 1 for sales…”)
• Conference rooms
• Call recording
• Ring groups (call multiple phones simultaneously)

VPS Requirements for VoIP

💡 Important: VoIP requires low latency. Choose a VPS location close to your team. For US-based teams, VPS.DO’s USA VPS is ideal. For Asia-Pacific teams, use Hong Kong VPS. View Plans →

Step 1: Prepare Ubuntu VPS

sudo apt update && sudo apt upgrade -y

# FreePBX works best on a minimal install
# Disable AppArmor (FreePBX requires this)
sudo systemctl stop apparmor
sudo systemctl disable apparmor

# Set hostname
sudo hostnamectl set-hostname pbx.yourdomain.com

Step 2: Install Dependencies

sudo apt install -y \
  build-essential git curl wget \
  apache2 mariadb-server mariadb-client \
  php8.1 php8.1-cli php8.1-mysql php8.1-curl \
  php8.1-mbstring php8.1-xml php8.1-zip \
  libapache2-mod-php8.1 \
  sox libsox-fmt-mp3 \
  ffmpeg lame mpg123 \
  nodejs npm \
  unixodbc libodbc1

sudo systemctl enable apache2 mariadb
sudo mysql_secure_installation

Step 3: Install Asterisk

cd /usr/src

# Download Asterisk 20 LTS
wget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20-current.tar.gz
tar xvfz asterisk-20-current.tar.gz
cd asterisk-20*/

# Install prerequisites
sudo contrib/scripts/install_prereq install

# Configure and compile
./configure --with-jansson-bundled
make menuselect  # Select desired modules (accept defaults for basic setup)
make -j$(nproc)
sudo make install
sudo make config
sudo make install-logrotation

# Start Asterisk
sudo systemctl enable asterisk
sudo systemctl start asterisk

# Verify
sudo asterisk -rx "core show version"

Step 4: Install FreePBX

cd /usr/src

# Download FreePBX
wget http://mirror.freepbx.org/modules/packages/freepbx/freepbx-17.0-latest.tgz
tar xvfz freepbx-*.tgz
cd freepbx/

# Create asterisk user
sudo useradd -m asterisk
sudo chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk \
    /var/log/asterisk /var/spool/asterisk /usr/lib/asterisk

# Run FreePBX installer
sudo ./install --webroot=/var/www/html \
    --dbengine=mariadb \
    --dbname=freepbx \
    --cdrdbname=asteriskcdrdb

# Enable FreePBX service
sudo systemctl enable freepbx
sudo fwconsole start

Step 5: Configure Apache for FreePBX

sudo nano /etc/apache2/sites-available/freepbx.conf

    ServerName pbx.yourdomain.com
    DocumentRoot /var/www/html
    
        Options -Indexes
        AllowOverride All
        Require all granted
    

sudo a2ensite freepbx
sudo a2enmod rewrite
sudo systemctl restart apache2

Access FreePBX at http://YOUR_VPS_IP/admin to complete the web-based setup. ✅

Step 6: Create SIP Extensions

In the FreePBX web interface:

1. Go to Applications → Extensions → Add Extension
2. Select PJSIP Extension
3. Configure:
   • User Extension: 100
   • Display Name: Alice Smith
   • Secret: (auto-generated secure password)
4. Click Submit → Apply Config

Repeat for each team member (101, 102, 103…).

Connect a SIP softphone

Use any SIP client (Zoiper, Linphone, 3CX mobile app) with these settings:

Step 7: Add a SIP Trunk (Real Phone Numbers)

To make and receive calls to/from regular phone numbers, you need a SIP trunk provider. Popular options: Twilio, VoIP.ms, Telnyx, Vonage.

Configure a SIP trunk in FreePBX

1. Connectivity → Trunks → Add Trunk → PJSIP Trunk
2. Trunk Name: MyProvider
3. Under PJSIP Settings:
   • Username: your SIP provider username
   • Password: your SIP provider password
   • SIP Server: sip.yourprovider.com

Set up outbound route

1. Connectivity → Outbound Routes → Add Route
2. Route Name: International Calls
3. Trunk Sequence: MyProvider
4. Dial Pattern: NXXNXXXXXX (matches US 10-digit numbers)

Set up inbound route

1. Connectivity → Inbound Routes → Add Route
2. DID Number: your purchased phone number
3. Set Destination: Extension 100 (or IVR, ring group, etc.)

Step 8: Configure Voicemail

1. Edit any extension → Voicemail tab
2. Enable voicemail: Yes
3. Voicemail password: 1234 (user can change)
4. Email address: alice@yourcompany.com
5. Email attachment: Yes (receive voicemail as audio file)

Configure email delivery in Admin → System Admin → Notifications.

Step 9: Security Hardening (Critical for VoIP)

VoIP servers are actively targeted by attackers who hijack them to make expensive international calls. This section is not optional.

Firewall rules

# Allow SIP (UDP/TCP)
sudo ufw allow 5060/udp
sudo ufw allow 5060/tcp
sudo ufw allow 5061/tcp  # SIP TLS

# Allow RTP media streams
sudo ufw allow 10000:20000/udp

# Allow HTTP/HTTPS for FreePBX admin
sudo ufw allow 80/tcp
sudo ufw allow 443/tcp

# RESTRICT admin access to your IP only
sudo ufw allow from YOUR_OFFICE_IP to any port 80
sudo ufw allow from YOUR_OFFICE_IP to any port 443
sudo ufw delete allow 80/tcp  # Remove unrestricted rules after adding IP-restricted ones

sudo ufw reload

Enable Fail2ban for Asterisk

sudo apt install fail2ban -y

sudo nano /etc/fail2ban/jail.local

[asterisk]
enabled  = true
filter   = asterisk
logpath  = /var/log/asterisk/messages
maxretry = 5
bantime  = 86400
findtime = 3600

sudo systemctl restart fail2ban

Restrict SIP access by IP

In FreePBX → Admin → Firewall, enable the FreePBX Firewall module and whitelist only your office/team IP ranges for SIP access.

Enable SIP TLS

In FreePBX → Admin → SIP Settings → TLS/SRTP, enable TLS for encrypted SIP signaling. Generate a certificate with Certbot and configure Asterisk to use it.

Monitor for fraud

# Watch for unusual call volume or destinations
sudo asterisk -rx "core show channels"
sudo asterisk -rx "sip show peers"

# Check CDR (call detail records) for anomalies
sudo mysql -u root -e "SELECT src, dst, duration, billsec FROM asteriskcdrdb.cdr ORDER BY calldate DESC LIMIT 20;"

Step 10: SSL for FreePBX Admin

sudo apt install certbot python3-certbot-apache -y
sudo certbot --apache -d pbx.yourdomain.com

Access your FreePBX admin interface securely at https://pbx.yourdomain.com/admin. ✅

Cost Comparison: Self-Hosted vs Hosted PBX

• 🇺🇸 USA VPS — from $20/month (low latency for US teams)
• 🇭🇰 Hong Kong VPS — for Asia-Pacific teams