Installing FreePBX 17 with Asterisk 22 LTS on Ubuntu 24.04 / 22.04

FreePBX is an open-source web-based GUI for managing Asterisk, the most widely deployed open-source PBX platform. It handles call routing, extensions, voicemail, IVR menus, and trunking through a clean web interface – no need to hand-edit Asterisk config files. FreePBX 17 is the latest GA release with support for Asterisk 22 LTS, PHP 8.2+, and a modernized module framework.

Original content from computingforgeeks.com - post 3161

This guide walks through building Asterisk 22 LTS from source (the latest long-term support release backed by security and bug fixes until 2029) and installing FreePBX 17 on top of it on Ubuntu 24.04 LTS. The same steps work on Ubuntu 22.04 with one adjustment noted in the PHP section. While FreePBX officially supports only Debian 12, this Ubuntu installation is fully functional – every command in this guide was tested on a live server.

Prerequisites

Before starting, make sure you have the following in place:

• Ubuntu 24.04 or 22.04 LTS server (dedicated – FreePBX manages its own Apache instance)
• 2+ vCPUs, 2+ GB RAM, 20+ GB SSD storage
• Root or sudo access
• A static IP address configured on the server
• Ports to open: 5060/UDP+TCP (SIP), 5061/TCP (TLS SIP), 10000-20000/UDP (RTP media), 80/TCP and 443/TCP (web admin)

Step 1: Update the System

Start by updating all packages to their latest versions and rebooting to apply any kernel updates.

sudo apt update && sudo apt upgrade -y
sudo reboot

Step 2: Install Asterisk 22 LTS Build Dependencies

Asterisk 22 LTS needs several development libraries and build tools to compile from source. Install them all in one shot.

sudo apt install -y build-essential git curl wget libnewt-dev libssl-dev libncurses5-dev \
  subversion libsqlite3-dev libjansson-dev libxml2-dev uuid-dev libsrtp2-dev \
  libspeexdsp-dev libedit-dev libopus-dev libvorbis-dev unixodbc-dev libpq-dev \
  automake autoconf libtool-bin sox mpg123 xmlstarlet ffmpeg pkg-config

Step 3: Download and Compile Asterisk 22 LTS

Download the latest Asterisk 22 LTS tarball from the official mirror and extract it.

cd /usr/src
sudo wget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22-current.tar.gz
sudo tar xzf asterisk-22-current.tar.gz
cd asterisk-22*/

Run the prerequisite installer script to pull in any remaining dependencies that Asterisk needs.

sudo contrib/scripts/install_prereq install

Once prerequisites are satisfied, configure the build with bundled pjproject and jansson libraries. These bundled versions avoid compatibility issues with system-packaged versions.

sudo ./configure --with-jansson-bundled --with-pjproject-bundled

Now compile Asterisk. The -j$(nproc) flag uses all available CPU cores to speed up the build.

sudo make menuselect.makeopts
sudo make -j$(nproc)
sudo make install
sudo make config
sudo ldconfig

Verify that Asterisk installed correctly by checking the version.

asterisk -V

You should see the version string confirming Asterisk 22 LTS:

Asterisk 22.8.2

Step 4: Create the Asterisk System User

Asterisk should run under its own dedicated user account rather than root. Create the asterisk user and group, then set ownership on all Asterisk directories.

sudo groupadd asterisk
sudo useradd -r -d /var/lib/asterisk -g asterisk asterisk
sudo usermod -aG audio,dialout asterisk
sudo chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/log/asterisk /var/spool/asterisk /usr/lib/asterisk

Install the sample configuration files so Asterisk has a working base config to start from.

cd /usr/src/asterisk-22*/
sudo make samples
sudo chown -R asterisk:asterisk /etc/asterisk

Edit /etc/asterisk/asterisk.conf to tell Asterisk to run as the asterisk user.

sudo sed -i 's/^;runuser = .*/runuser = asterisk/' /etc/asterisk/asterisk.conf
sudo sed -i 's/^;rungroup = .*/rungroup = asterisk/' /etc/asterisk/asterisk.conf

Start Asterisk to confirm it launches without errors.

sudo -u asterisk /usr/sbin/asterisk

Step 5: Install Apache, MariaDB, PHP 8.3, and Node.js 22

FreePBX needs a web server, database, PHP, and Node.js. Ubuntu 24.04 ships PHP 8.3 in its default repos, which meets FreePBX 17’s requirement of PHP 8.2 or higher.

sudo apt install -y apache2 mariadb-server mariadb-client \
  php8.3 libapache2-mod-php8.3 php8.3-mysql php8.3-curl php8.3-cli \
  php8.3-zip php8.3-xml php8.3-gd php8.3-mbstring php8.3-intl \
  php8.3-bcmath php8.3-soap php8.3-ldap php8.3-sqlite3 php8.3-imap \
  php8.3-common unixodbc

Ubuntu 22.04 note: The default PHP version on Ubuntu 22.04 is 8.1, which is too old for FreePBX 17. You need to add the ondrej/php PPA first to get PHP 8.3:

sudo add-apt-repository ppa:ondrej/php -y
sudo apt update

Then install the same php8.3-* packages listed above.

Install Node.js 22

FreePBX 17 requires Node.js for its UCP (User Control Panel) and other frontend components. Install Node.js 22 from the NodeSource repository.

curl -fsSL https://deb.nodesource.com/setup_22.x | sudo -E bash -
sudo apt install -y nodejs

Verify the installed versions match what FreePBX expects.

php -v
node -v
mariadb --version

Expected output should show PHP 8.3.6, Node.js v22.22.1, and MariaDB 10.11.14.

Configure Apache for FreePBX

FreePBX needs Apache to run as the asterisk user (same user that owns all Asterisk files) and requires mod_rewrite enabled for its .htaccess rules.

sudo a2enmod rewrite
sudo sed -i 's/^export APACHE_RUN_USER=.*/export APACHE_RUN_USER=asterisk/' /etc/apache2/envvars
sudo sed -i 's/^export APACHE_RUN_GROUP=.*/export APACHE_RUN_GROUP=asterisk/' /etc/apache2/envvars
sudo usermod -aG www-data asterisk

Enable AllowOverride All for the web root so FreePBX’s .htaccess files work properly.

sudo sed -i '/<Directory \/var\/www\/>/,/<\/Directory>/ s/AllowOverride None/AllowOverride All/' /etc/apache2/apache2.conf
sudo systemctl restart apache2

Step 6: Download and Install FreePBX 17

Download the latest FreePBX 17 release package from the official mirror and extract it.

cd /tmp
wget http://mirror.freepbx.org/modules/packages/freepbx/freepbx-17.0-latest.tgz
tar xzf freepbx-17.0-latest.tgz
cd freepbx

Run the FreePBX installer. The -n flag runs a non-interactive install using the default MariaDB root credentials.

sudo ./install -n --dbuser root --dbpass ''

The installer connects to MariaDB, creates the FreePBX databases, deploys the web files to /var/www/html, and registers the Asterisk connection. A successful install ends with this message:

You have successfully installed FreePBX

Now install all available FreePBX modules and reload the configuration. This pulls in voicemail, call recording, IVR, ring groups, and dozens of other modules.

sudo fwconsole ma installall
sudo fwconsole reload

Step 7: Enable FreePBX as a Systemd Service

Create a systemd unit file so FreePBX starts automatically on boot and can be managed with standard systemctl commands.

echo '[Unit]
Description=FreePBX VoIP Server
After=mariadb.service

[Service]
Type=oneshot
RemainAfterExit=yes
ExecStart=/usr/sbin/fwconsole start -q
ExecStop=/usr/sbin/fwconsole stop -q

[Install]
WantedBy=multi-user.target' | sudo tee /etc/systemd/system/freepbx.service

Enable and start the service.

sudo systemctl daemon-reload
sudo systemctl enable freepbx
sudo systemctl start freepbx

Check the service status to confirm it started without errors.

sudo systemctl status freepbx

Step 8: Configure the Firewall

Open the required ports for SIP signaling, RTP media streams, and web administration. If you plan to use SIP over TLS, port 5061 is needed as well.

sudo ufw allow 80/tcp
sudo ufw allow 443/tcp
sudo ufw allow 5060/udp
sudo ufw allow 5060/tcp
sudo ufw allow 5061/tcp
sudo ufw allow 10000:20000/udp
sudo ufw enable

Verify the firewall rules are active.

sudo ufw status

The following table summarizes all ports FreePBX and Asterisk need:

Port	Protocol	Purpose
80	TCP	FreePBX web admin (HTTP)
443	TCP	FreePBX web admin (HTTPS)
5060	UDP/TCP	SIP signaling
5061	TCP	SIP over TLS
10000-20000	UDP	RTP media streams

Step 9: Access the FreePBX Admin Console

Open a browser and navigate to http://203.0.113.50/admin (replace with your server’s IP address). The first time you access the admin panel, FreePBX prompts you to create an administrator account. Set a strong username, password, and notification email address.

After creating the admin account, you land on the FreePBX dashboard. This is the central hub showing system status, Asterisk connection health, and module update notifications.

The Module Admin page (Admin > Module Admin) shows all installed modules and their versions. After running fwconsole ma installall earlier, you should see core, framework, voicemail, IVR, ring groups, and many other modules listed as enabled.

The Asterisk Info page (Admin > Asterisk Info) displays detailed information about the running Asterisk instance, including the version, uptime, loaded modules, and active channels.

Step 10: Create SIP Extensions

With FreePBX running, the next step is creating SIP extensions so phones and softphones can register. FreePBX 17 uses PJSIP as the default channel driver (chan_pjsip), which is the modern replacement for the legacy chan_sip.

To create a new extension, go to Applications > Extensions. You will see the extensions list page.

Click Add Extension and select Add New PJSIP Extension. Fill in the required fields:

• User Extension – the extension number (e.g., 1001)
• Display Name – the caller ID name shown on other phones
• Secret – the SIP password for phone registration (auto-generated or set your own)

Click Submit and then Apply Config (the red bar at the top) to write the changes to Asterisk. The extension is now ready for phone registration. Configure your SIP phone or softphone with the server IP (203.0.113.50), extension number, and secret to start making calls.

Step 11: Verify the Installation

Confirm FreePBX and Asterisk are running with the expected versions.

sudo fwconsole --version

This should report the FreePBX framework version:

FW Console - FreePBX Utility 17.0.19.32

Check the running Asterisk version through the Asterisk CLI.

sudo asterisk -rx 'core show version'

The output confirms Asterisk 22 LTS is active:

Asterisk 22.8.2 built by root on a x86_64 running Linux on 2026-03-22 18:30:15 UTC

You can also verify all services are running and listening on the expected ports.

sudo ss -tlnp | grep -E ':(80|5060|5061) '

Conclusion

You now have FreePBX 17 with Asterisk 22 LTS running on Ubuntu 24.04. The system is ready for extension creation, trunk configuration, and inbound/outbound call routing through the web admin panel at http://203.0.113.50/admin.

For a production deployment, there are a few additional steps to harden the system. Set up Let’s Encrypt SSL certificates using certbot to encrypt the web admin and SIP TLS traffic. Install fail2ban with the Asterisk jail to block SIP brute-force attacks – this is critical for any internet-facing PBX. Set up automated backups using fwconsole backup and store them off-server. The FreePBX wiki has detailed documentation on advanced configuration including trunking with SIP providers, IVR setup, and alternative PBX platforms if FreePBX does not fit your needs.