Deploy VoIP Services with Asterisk and Freepbx on Ubuntu 12.04
Updated by Alex Fornuto
Asterisk is an open-source telephone solution that runs over the Internet instead of running through copper lines. It offers a variety of features such as voicemail and conference calling, much like a landline telephone can.
For this guide we will install Asterisk from source rather than from Ubuntu’s repositories. The newer version offers several additional features, including the ability to integrate a Google Voice account as a trunk. We will use FreePBX as a web interface for our Asterisk configuration.
Please note: Because of the special configuration options required for this setup, you should not run other services on the Linode you intend to use Asterisk on. It is also worth noting that this guide will walk you through using PV-GRUB. Any alterations to the steps in this guide will fall outside the scope of support.
The steps required in this guide require root privileges. Be sure to run the steps below as
rootor with the sudo prefix. For more information on privileges see our Users and Groups guide.
If you’re new to the idea of running a digital phone system, you may want to brush up on some of the protocols and acronyms used most frequently.
- PBX Private Branch Exchange - A system that provides telephone switching and connection for an internal system. Asterisk with the FreePBX front-end will serve as our PBX.
- Trunk - A service provider that will deliver phone calls to and from your PBX platform. There are many providers offering a variety of solutions. In this guide we’ll cover using Google Voice as your trunk, but you can replace those instructions with those provided by another trunk provider to complete your setup.
- DID Direct Inward Dialing number - Usually assigned by the trunk provider, this is the phone number (or numbers) used by the outside world to call into your PBX system.
- SIP Session Initiation Protocol - This protocol works over Internet Protocol (IP) to establish multimedia connections. Our PBX server will use SIP to communicate with the trunk provider as well as the client device.
The diagram below shows the relationship between each of the components that allow a standard phone and your SIP client to communicate:
Before you begin, you need to make sure a few things are in order. We assume you have followed the Getting Started guide and have set the hostname and timezone, and have configured networking for the Linode. These last steps are of particular importance for ensuring your Asterisk installation functions normally. If you plan on using Asterisk’s email features, you may also wish to add an A record for your domain.
There are quite a few prerequisites to satisfy before you can begin installing Asterisk and FreePBX. Most notably, you will need to install a kernel module and change your Linode’s configuration profile. We’re going to outline the instructions for doing so in this document. If you want a more detailed explanation, you may wish to take a look at the in-depth information contained in the PV-GRUB guide.
This guide includes instructions for integrating a Google Voice account. You will need a Google account that’s already configured with a Google Voice number to complete these steps.
Install Required System Packages
Issue the following commands to update your package lists and upgrade packages on your Linode:
apt-get update apt-get upgrade
Install the the set of programs below, which we will need later to build the Asterisk software from source:
apt-get install build-essential wget libssl-dev libncurses5-dev libnewt-dev libxml2-dev libsqlite3-dev uuid-dev libiksemel-utils libiksemel-dev
Create Asterisk User
We’re going to create a user to run the Asterisk process, so that we’re not running everything as root. Issue the following command:
You will be prompted for a password and some details for the user such as name and phone number. You need to fill out the password, but you may safely hit
Enter for the other entries.
Install the Kernel
You will need to use the PV-GRUB kernel provided by Linode. This method works, but any kernel problems that arise from editing the kernel beyond the steps outlined in this document will not be supported by Linode. You’ll need to prepare your Linode before updating your configuration profile by following the commands in the next section.
Asterisk uses the dahdi_dummy kernel module, which requires you to edit a few things in the kernel as well as your Linode’s configuration profile. This is a straightforward process; issue the following command:
apt-get install linux-virtual
You will be presented with a menu like the one below asking you to choose a disk to install GRUB on. You can continue without choosing an option, since this version will be purged in the next step.
Install and Configure Grub
GRUB is a bootloader that will allow you to boot the kernel we’re setting up. You will need to install and create a default configuration file for it. Issue the following commands:
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apt-get purge grub2 grub-pc apt-get install grub update-grub
You will receive a warning about the
/boot/grub/menu.lstfile not existing and be asked if you would like to create one. Enter y to create a new file.
/boot/grub/menu.lstat the “groot” line. Please note: Directives in this file look as if they are commented out. Leave the hash marks (“#”) as they are.
grootline to resemble the following:
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## default grub root device ## e.g. groot=(hd0,0) # groot=(hd0)
indomUline to resemble the following:
Save and exit the file. You will now need to update GRUB again in order to apply the changes. Issue the following command:
Edit Configuration Profile
You will now need to log in to the Linode Manager in order to change your Linode’s configuration profile.
- Navigate to the Dashboard page of the Linode you are going to use for Asterisk.
- Click the profile you are currently using and select pv-grub-x86_64 (or pv-grub-x86_32 if you are using a 32 bit system) from the kernel drop down.
- Save this configuration profile. You may wish to change its name to indicate that this is no longer a default profile.
- Reboot your system to make sure that these changes are applied. You will need to do this before you can proceed. It is a good idea to watch the shutdown and reboot phases via LISH to see if there are any errors.
It’s very important that you follow the steps outlined above carefully or your system may not boot. It is highly recommended that you watch the console during the shutdown and reboot phases via LISH. If your Linode does not boot and you get an error, change your configuration profile back to the latest Paravirt kernel and read over this guide to make sure you have not missed any steps.
The version of Asterisk in the Ubuntu repository is outdated and can’t support all the features needed for this guide, so begin by downloading the current build of Asterisk 11 directly. You will also need to download the libpri library and the DAHDI module:
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cd /usr/src/ wget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gz wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
Extract the files:
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tar zxvf dahdi* tar zxvf libpri* tar zxvf asterisk*
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cd /usr/src/dahdi-linux-complete* make make install make config
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cd /usr/src/libpri* make make install
Begin Building Asterisk:
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cd /usr/src/asterisk* ./configure make menuselect
This will bring up a menu of additional modules to install. Make sure that under
Now go up to
Channel Driversand select
When you’ve selected these and any other modules or build options you may want, use the Save & Exit button. Then issue the following commands to finish the installation:
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make make install make config make samples
Start the DAHDI and Asterisk services:
service dahdi start service asterisk start
Verify the Installation
To make sure all the required packages are installed properly, launch the Asterisk CLI:
The output should be similar to the excerpt below:
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Asterisk 11.8.1, Copyright (C) 1999 - 2013 Digium, Inc. and others. Created by Mark Spencer <email@example.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 11.8.1 currently running on localhost (pid = 6963) localhost*CLI>
The following commands and output serve to verify that DAHDI and libpri are properly installed:
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localhost*CLI> dahdi show version DAHDI Version: 2.9.0 Echo Canceller: localhost*CLI> pri show version libpri version: 1.4.14
FreePBX is a PHP application that allows you to control your Asterisk installation through a web interface.
Set Up LAMP Stack
Before you can use FreePBX, you will need to set up a LAMP stack. An basic step-by-step how-to is provided here, but you may wish to consult our LAMP documentation for more information.
Begin by installing Apache:
apt-get install apache2
apt-get install mysql-server
MySQL will ask you to set a root user password. Remember what you choose, as you will need to use it again later on.
Secure the default installation by issuing the following command:
Assuming you set a strong root user password in the previous step, you can skip the first option to change the root password. For all other suggestions, choose Y.
Create the databases to be used by the Asterisk system. Start by launching the MySQL CLI:
You will be asked for the password you created when you installed MySQL, then you will be taken to the MySQL command line interface.
Run the following commands, in this order. Be sure to change the password (CHANGEME) to something secure that you will remember. These commands will create two databases named asterisk and astieriskcdr, and give the
asterisksystem user permission to access and modify their contents.
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create database asterisk; create database asteriskcdr; GRANT ALL PRIVILEGES ON asteriskcdr.* TO asterisk@localhost IDENTIFIED BY 'CHANGEME'; GRANT ALL PRIVILEGES ON asterisk.* TO asterisk@localhost IDENTIFIED BY 'CHANGEME'; flush privileges; exit
Install PHP and related packages:
apt-get install php5 php-pear php5-mysql php5-suhosin php5-cgi php-pear php-db libapache2-mod-php5
For this installation, we want Apache to run as the Asterisk user. This will allow Apache to access all of the files it needs in order to run FreePBX. Make sure your
/etc/apache2/envvarsfile resembles the following:
export APACHE_RUN_USER=asterisk export APACHE_RUN_GROUP=asterisk
Change ownership of the Apache lock file:
chown asterisk:asterisk /var/lock/apache2/
Finally, restart Apache to make sure everything is loaded correctly:
service apache2 restart
Download and Extract FreePBX
Obtain FreePBX and unpack it by issuing the following commands:
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cd /opt wget http://mirror.freepbx.org/freepbx-188.8.131.52.tgz tar -xvf freepbx-184.108.40.206.tgz chown -R asterisk:asterisk freepbx* su - asterisk cd /opt/freepbx/
This is the latest version of FreePBX at the time of this guide’s publication. You can check this page for the latest build of FreePBX.
The FreePBX directory contains SQL files that you can insert into the database we created previously. Issue the following commands to insert this data:
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mysql -p asterisk < SQL/newinstall.sql mysql -p asteriskcdr < SQL/cdr_mysql_table.sql exit
The FreePBX configuration process will ask you a series of questions that you should pay attention to. Some of the default values are fine for a production system, but you will want to change the passwords to prevent unauthorized access.
You need to pass the credentials of the MySQL user and database you created above to the configuration script. Issue the following command:
cd /opt/freepbx/ ./install_amp
Before you continue your FreePBX installation, you will want to configure a
VirtualHost for the web interface. You should also secure your installation using SSL and an
.htaccess file. By default, FreePBX installs files to
/var/www/html/; you may leave this as it is. Your
VirtualHost may resemble the following:
- VirtualHost Entry
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<VirtualHost *:80> ServerAdmin firstname.lastname@example.org ServerName example.com ServerAlias www.example.com DocumentRoot /var/www/html </VirtualHost>
To update your Apache configuration, you will need to restart the server. Issue the following command:
service apache2 reload
You should now be able to visit your Linode’s IP address or the domain you’ve pointed at your Linode in your web browser. The first page should prompt you to create a user and password to administer the system from:
Now we will use the FreePBX web interface to configure your phone server.
Update the Modules
From your web browser, go to your FreePBX web interface.
Under the Admin menu select Module Admin.
Select the Unsupported repository, then press the Check Online button.
- Select the Upgrade All link. Under the Connectivity section, select Google Voice/Chan Motif. If there are other modules on this list you want to install, select them now.
When finished, press the Process button, and then Confirm.
If downloading modules fails, from the terminal run this command from the terminal:
su asterisk -c 'mkdir /var/www/html/admin/modules/_cache'. This should resolve the issue.
You will now see a red Apply Config button. Use it to enable the modules and updates you just downloaded.
To prevent a bug later on, go back to your terminal and issue the following commands:
cd /etc/asterisk rm ccss.conf confbridge.conf features.conf sip.conf iax.conf logger.conf extensions.conf sip_notify.conf
Then repeat Steps 1-4* but select instead to install or reinstall the
Create an Extension
Your PBX system will need at least one defined extension to send incoming calls to. Note that you can repeat this section for additional extensions.
- Under the Applications menu select Extensions.
Unless you have a custom device to configure, keep the drop down menu at
Generic SIP Deviceand hit submit.
Create your SIP extension. The only fields required are the Extension itself, a Display Name, and a secret (password).
Configure Integration with Google Voice
The steps in this section will let your Asterisk server use Google Voice as a trunk for sending and receiving phone calls.
From the FreePBX web interface, under the Connectivity menu select Google Voice (Motif). Fill out your Google Voice account information and check the options as show below:
Under Connectivity go to Inbound Routes. Create an inbound route pointing to your extension:
Now you can point your SIP client at your Linode, sign in with your extension and password, and begin making and receiving calls.
You may wish to consult the following resources for additional information on this topic. While these are provided in the hope that they will be useful, please note that we cannot vouch for the accuracy or timeliness of externally hosted materials.
This guide is published under a CC BY-ND 4.0 license.