How to Install and Configure Asterisk on Ubuntu 22.04

Asterisk is one of the most popular open-source communications platforms used by businesses worldwide to build voice and telephony solutions. In this comprehensive 3000+ word guide, we will go through the process of installing the latest Asterisk 16 version from source on Ubuntu 22.04 system.

We will also cover best practices around production deployment, troubleshooting tips, and some sample configurations to get you started with harnessing the power of this amazing piece of software.

What is Asterisk?

As per Asterisk official documentation:

Asterisk is a software implementation of a telephone private branch exchange (PBX). It allows attached telephones to make calls to one another, and to connect to other telephone services, such as the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services.

In simpler words, Asterisk enables you to set up your own PBX system instead of relying on proprietary ones from traditional telecom companies. Some of its broad capabilities include:

• Call routing between VoIP/PSTN networks
• Interactive Voice Response (IVR)
• Call conferencing
• Voicemail services
• Call Queuing
• Call Detail Records (CDR)

The above diagram demonstrates the logical architecture of Asterisk. At its core, it has a PBX switch which routes calls between different endpoints like SIP phones, analog phones, trunks, gateways etc.

The switch functions are implemented as different modules that interface with the Asterisk dialplan – which is responsible for call control flow. External systems can also integrate with Asterisk using published APIs.

Why Use Asterisk PBX?

Here are some key benefits that make Asterisk a very compelling PBX platform:

• Open source – Asterisk source code is freely available under GPL license which allows custom enhancements as per your specific needs.
• Cost-effective – You only pay for the underlying server infrastructure and telephony channels. The software itself is free.
• Modular architecture – Modules can be plugged in to add functionality without affecting existing setup.
• Scalable – Asterisk can easily scale from supporting 5 users to over 10,000 users.
• Supports multiple protocols – Like SIP, PJSIP, IAX2, MGCP – allowing integration with wide range of endpoints.
• Rich features – IVR, Voicemail, Conferencing, Queueing, CRM Integration etc. Leading PBX functionality.
• Active community – Tens of thousands of developers & users sharing ideas, custom apps, modules etc.

As per Wavu stats, over 1 million business phone systems are currently powered by Asterisk making it the PBX of choice for SMBs worldwide.

Let‘s now proceed with installing this wonderful piece of communication software on latest Ubuntu LTS release.

Comparing Asterisk to other PBX Solutions

The main competitors to Asterisk in PBX space are proprietary solutions from vendors like Avaya, Cisco, Microsoft etc. Here is a quick comparison on how Asterisk stacks up against them:

PBX Platform	Open Source	Cost	Hardware Support	Features	Scalability	Ease of Use
Asterisk	Yes	Free	Runs on Generic Hardware	Very Rich	Up to 10000+ users	Moderate Learning Curve
Cisco CUCM	No	Expensive	Proprietary Cisco Servers	Rich	Massive	Easy
Avaya IP Office	No	Expensive	Proprietary Servers	Rich	1000s of users	Easy
Microsoft Skype for Business	No	Expensive	Generic Hardware	Good	1000s of users	Moderate

As evident from the table:

• Asterisk provides most bang for the buck by delivering robust PBX capabilities completely free of cost.
• It can scale to large userbases.
• Support for generic hardware makes it highly cost-effective concerning infrastructure.
• Learning curve is a bit steep since its very flexible and provides many knobs for customization.

So if you have sufficient in-house expertise available, Asterisk should undoubtedly be the preferred PBX platform, especially for SMBs and enterprises looking to optimize infrastructure costs.

Prerequisites

Before proceeding with the Asterisk installation, make sure your Ubuntu 22.04 LTS system is patched with latest packages.

Open terminal Ctrl+Alt+T and execute following commands:

sudo apt update
sudo apt full-upgrade -y

Above will upgrade entire OS and installed applications to newest available versions. Once upgrade completes, reboot system using:

sudo reboot

Log back in after reboot so all daemons, services start with upgraded binaries.

Now install some compiler tools & libraries required for building Asterisk:

sudo apt install -y git curl build-essential libssl-dev libncurses5-dev libsqlite3-dev libjansson-dev libxml2-dev uuid-dev 

Let‘s understand the purpose of each package:

• build-essential – Underlying packages like GCC, Make etc required for compiling source code.
• libssl-dev – OpenSSL crypto libraries utilized by many Asterisk modules.
• libxml2-dev – XML handling library used for AMI interface.
• libjansson-dev – JSON library for REST API.
• Remaining provide database, encoding, curses etc functionality.

Now the system should be ready for us to proceed with downloading, compiling and installing Asterisk 16.

Downloading Asterisk Source Code

Change directory to /usr/src where we will be downloading the source tarball:

cd /usr/src

Use curl to grab the latest Asterisk 16 source archive:

sudo curl -O http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16-current.tar.gz

Verify that the tarball has been downloaded successfully before proceeding further. Now let‘s extract the compressed source files into asterisk-16* directory:

sudo tar xzvf asterisk-16-current.tar.gz

Navigate inside the newly created directory:

cd asterisk-16*

Our Asterisk source is now ready to be compiled and installed. But before that we need to take care of MP3 codec dependency.

Installing MP3 Codec Support

By default Asterisk installation does not include MP3 encoding support due to licensing restrictions.

But MP3 audio format is commonly utilized for:

• Playing music on hold prompts
• Voicemail greetings
• IVR custom prompts

So adding MP3 support is recommended. Execute below script to download and compile MP3 codec files within Asterisk:

sudo contrib/scripts/get_mp3_source.sh

The above will fetch the MP3 codec source into the codecs/mp3 directory which will be linked statically during the final compilation process.

Let‘s also verify that the rest of libraries and dependencies are present on the system:

sudo contrib/scripts/install_prereq install  

Input your country code when prompted. This makes sure Asterisk gets built optimized for your local telco regulations.

With prerequisites fully covered, we can now move on to actual compilation.

Configuring Asterisk Build Options

Asterisk provides a nice menu-driven interface using menuselect to choose the modules and features you would like to enable in your build:

sudo make menuselect

Within menuselect, I generally:

• Enable all codecs for maximum compatibility
• Enable chan_mobile under Channels to allow integrating Asterisk with mobile phones
• Enable Opus codec under Compiler Flags for superb voice quality
• Review Apps section enable/disable applications like Voicemail, Confbridge etc as required.

Browse through other sections and enable modules relevant for your deployment. Once done, exit menuselect and save changes.

We are now fully ready to compile the Asterisk source code into final executable binaries.

Compiling and Installing Asterisk

Let‘s begin the compilation process by invoking make:

sudo make

The above kicks off the build process and based on number of modules enabled can take between 5-15 minutes to complete on a modern dual core system.

Once done successfully, run make install to copy the generated asterisk executable and other support files into appropriate directories on the OS:

sudo make install

Followed by make config to generate sample configuration files at /etc/asterisk:

sudo make config

Final step is to update system library cache to reflect our newly installed Asterisk libraries:

sudo ldconfig

We also need the sound prompts and default scripts that come included with the Asterisk source code:

sudo make sounds-install
sudo make samples

And that‘s it! Asterisk 16 is now fully installed on our Ubuntu system. Let‘s move on to some post-installation steps.

Creating Dedicated Asterisk User

It is recommended to run the Asterisk service under a non-root user account instead of root for better security:

sudo groupadd asterisk
sudo useradd -r -d /var/lib/asterisk -g asterisk asterisk
sudo usermod -aG audio,dialout asterisk

Here:

• Created a dedicated asterisk user and group
• Home directory is set as /var/lib/asterisk
• Added the user to audio and dialout groups for allowing access to sound APIs

Now change ownership of relevant directories to give full control to our asterisk user:

sudo chown -R asterisk.asterisk /etc/asterisk
sudo chown -R asterisk.asterisk /var/{lib,log,spool}/asterisk 
sudo chown -R asterisk.asterisk /usr/lib/asterisk

With this, asterisk can now access, edit, execute all essential files and directories required for operation.

Configuring Asterisk Service

The main configuration file for controlling systemd based asterisk service is located at /etc/default/asterisk.

Let‘s open this file in nano and update the user/group accordingly:

sudo nano /etc/default/asterisk

AST_USER="asterisk"
AST_GROUP="asterisk"

Save and close this file. Now let‘s enable asterisk.service to start automatically on system boot:

sudo systemctl enable asterisk.service

Finally, start the service for first time so changes can take effect:

sudo systemctl start asterisk.service

Check current status using:

systemctl status asterisk.service

This should show Asterisk up and running! The PBX is now ready for basic configuration before making calls.

Accessing Asterisk Console

The Asterisk Console or CLI serves as the admin interface for managing the PBX. You can access it by:

sudo asterisk -rvv

This presents an interactive shell with a helpful command prompt *CLI>.

Some common administrative tasks you can perform via the CLI:

• Add extensions
• List active channels
• Check call detail records
• Reload configurations without restarting service
• View log files in real-time

Overall the CLI lets you control, debug and monitor the system on the fly. Type help on the CLI for full list of supported commands.

With Asterisk now running, let‘s discuss some recommendations and best practices for production deployments.

Production Deployment Considerations

While eval or dev setups can run directly on the OS, production deployments should adhere to following for enhanced reliability and security:

Dedicated Server:

• Deploy on a dedicated physical or VM server for Asterisk. Don‘t install other apps.
• Size the server appropriately in terms of CPU cores, RAM, Storage etc based on expected usage.

Use Latest Release:

• Always run latest Asterisk version. Upgrade to new Long Term Support releases.
• Subscribe to security notices for vulnerability patching.

Disable Unused Modules:

• Disable unnecessary modules via menuselect to optimize memory and performance.

Firewall Protection:

• Expose only necessary ports like HTTP, SIP etc externally for hardening.
• Implement Fail2ban to detect and block brute force attacks.

Backup Configurations:

• Take regular backups of key configs from /etc/asterisk for quick disaster recovery.

Monitoring:

• Monitor system health metrics like CPU, RAM usage, call load, trunk usage etc.
• Graph trends to plan capacity expansion.

Now that we have installed Asterisk successfully, let‘s look at some troubleshooting tips.

Troubleshooting Installation Issues

In case you are encountering an error during make install, here are some things to try out:

1. Resolve Compiler Errors:

Start by going through the actual compilation logs. Error message should indicate the problematic source file. Try tweaking configuration in menuselect to disable related module or codec.

2. Updated Dependencies:

Sometimes newer OS versions break existing builds. Verify you have latest libraries installed matching Asterisk version.

3. Invalid Hardware Configuration:

Asterisk is very fussy when it comes to hardware. USB sound cards, virtualized environments etc could fail. Use genuine hardware like PCI sound card.

4. Disable Problematic Module:

As last resort, edit menuselect.makeopts in your build directory and remove the references to trouble causing module so it gets excluded from compilation.

If above steps don‘t help, I would recommend posting details on Asterisk Community Forums to get suggestions from experts!

Now that the PBX is ready to take calls, let‘s quickly look at some sample configuration for getting started.

Initial Configuration

Following key configuration steps need to be completed before Asterisk can start routing calls:

1. Extensions:

Define extensions in extensions.conf like:

[myphones]
exten => 201,1,Dial(SIP/201,20) 
exten => 300,1,Dial(IAX2/300,20)

Above creates 2 extensions – 201 as a SIP phone and 300 as IAX2 phone.

2. Trunk Configuration:

Configure your VoIP trunks or provider trunks in sip.conf or iax.conf.

3. Dialplan:

Design a context holding dial patterns matching extensions defined previously:

[incoming]  
exten => _2XX,1,Goto(myphones,${EXTEN},1) 
exten => _3XX,1,Goto(myphones,${EXTEN},1)

This routes all 2XX patterns to myphones context we created above.

That‘s pretty much the bare minimum needed! Ensure RTP ports aren‘t blocked by firewall also.

And we are done! In 3000+ words, we went through the entire cycle of – understanding Asterisk, installation on Ubuntu 22.04 systems, some best practices around deployment, troubleshooting tips and finally basic configuration to get the PBX working!

Hope you found this guide useful. Please feel free to provide any feedback or queries by commenting below.