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Vaidas Jablonskis

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Asterisk is software that turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and more.

This howto covers topics and issues installing, configuring and managing Asterisk server on Fedora Linux version 12 to 14. It is based on Asterisk 1.6.2.


I assume you are going to be using Asterisk and its components packages which Fedora supplies via yum ‘updates’ repository, which is enabled by default.

Required packages for a simple Asterisk installation. Is is always a good practice to install sound files for the most common codecs, so asterisk’s core does not have to convert the sound files on the fly - asterisk-sounds-core-en-<codec>

asterisk-voicemail # (if you need voicemail support)
iax # (if you need IAX protocol support)

Packages installation using yum install:

yum install asterisk asterisk-sounds-core-en asterisk-sounds-core-en-alaw \
  asterisk-sounds-core-en-g722 asterisk-sounds-core-en-g729 \
  asterisk-sounds-core-en-gsm asterisk-sounds-core-en-ulaw \
  asterisk-sounds-core-en-wav asterisk-voicemail iax

Turn on asterisk service to start on boot

chkconfig --level 345 asterisk on


First of all let’s start the Asterisk service just to make sure it starts without any errors: service asterisk start

Once the service is started and running you can connect to Asterisk by simply running asterisk -r - the -v option tells the verbosity level of the Asterisk’s core (it can be set via Asterisk’s command line), which is optional. You will get into the Asterisk’a command prompt:

# asterisk -r
Asterisk, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer
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 currently running on zs (pid = 20283)
Verbosity is at least 10

The main components are the following:

  • core - Asterisk’s core engine
  • dialplan - Asterisk’s logic which you will have to define
  • sip - The SIP protocol engine
  • iax2 - The IAX2 protocol engine


#### Config files

The files are stored in /etc/asterisk directory. The main configuration files are:

  • sip.conf - This is where we’ll configure the SIP protocol
  • extensions.conf - A dialplan configuration file, this is where all the logic we define goes to
  • iax.conf - This is the IAX2 protocol configuration file

Backup existing files

Asterisk packaged installation comes with already populated config files, it has some good examples, but they are unrelated to what we are going to achieve, so it is a good idea to backup them using:

cd /etc/asterisk
mv sip.conf sip.conf.sample; touch sip.conf
mv iax.conf iax.conf.sample; touch iax.conf
mv extensions.conf extensions.conf.sample; touch extensions.conf

SIP Protocol

Since the SIP protocol is the most common one, I will cover its configuration in this howto. The sip.conf file contains channel and users (phones) configuration, login details etc. The main channel is [general] which is reserved for general/default SIP engine configuration. Many options can be set per channel or trunk channel. A new channel starts with [channel-name] and ends where the new channel starts.

The basic structure of the sip.conf file is:

;this is a comment
context=default ;if no destination is specified the call will go to the default context
bindport=5060 ;port number to bind (default)
bindaddr= ;address to bind (all by default)
srvlookup=yes ;do DNS lookups
directmedia=no ;use indirect media (partially solves SIP over NAT issues)
bandwidth=low ;enable most common codecs
disallow=lpc10 ;makes your voice sound like a robot, so we disable this codec :-)

;SIP provider registration
;register = username:[email protected]

;simple configuration for user (extension) 1001
type=friend ;type friend allows a user/phone to make and receive calls
host=dynamic ;host is needed when incoming calls comes in, Asterisk will take a note of phone's IP upon registration
context=users ;the context for inbound and outbound calls in this case (because type is friend)
callerid="Name Lastname <1001>"

Dial Plan

The main Asterisk PBX logic is configured in extensions.conf file. This file again is separated by [sections]. The already reserved sections/contexts are [globals] and [general]. [globals] is usually empty, but can contain some options or it can be used for variable assignments. Let’s create a basic extensions.conf configuration file:



[default] ;create a default context, so if incoming caller did not specify an extension the call will end up in this context
exten => s,1,Verbose(1,Unrouted call handler) ;send a custom message in the log
exten => s,n,Dial(SIP/1001) ;call extension 1001 (remember, we configured 1001 channel in sip.conf)
exten => s,n,Hangup() ;and finally hang up once the call is finished, it is always a safe bet to do that

[users] ;create a users context, this is where configured users/phones will end up (remember? we placed user 1001 in 'users' context)
exten => 1001,1,Dial(SIP/1001)
exten => 1001,n,Hangup()

exten => 5000,1,Echo() ;create a simple echo test on ext 5000
exten => 5000,n,Hangup()

So now you should have the working configuration files for SIP channels and a simple dialplan. Let’s move on to the Asterisk basic management and control.

Asterisk and NAT

Probably the worst case is when an Asterisk PBX is behind one NAT and clients connecting to the PBX are behind another NAT and network. Asterisk can handle this situation pretty well if you tell it to. There few steps which need to be done in order to make Asterisk property translate SIP and RTP data over NAT:

Do port forwarding on the firewall which does NAT for the Asterisk server of the following ports: 5060/udp (SIP traffic), 10000-20000/udp (RTP traffic) - ports range can be changed on /etc/asterisk/rtp.conf file.

Add the following options to /etc/asterisk/sip.conf file:


externip= - set it to your external IP
localnet= - set it to your local subnet (more than one 'localnet' option can be used)
qualify=yes - set it to 'yes', so Asterisk will keep connections open over the NAT
canreinvite=no - set it to no, so Asterisk does not send re-invites (always stays in between the current call)


Now we have configured basic components of the Asterisk PBX, but our PBX system is running off the sample configuration files, remember (we renamed the files after we have started the Asterisk service)? I believe you still have the terminal open with a connection to the service (asterisk -r). Asterisk control is pretty simple, I will list a few of the main commands just to get you started:

  • core set verbose 10 - increases/decreases verbosity level of the core (10 is the most verbose output)
  • core restart - restarts the Asterisk core, very rarely needed, unless you are changing core configuration
  • core show channels - displays active channels/calls and processed calls
  • sip set verbose 10 - increases/decreases verbosity level of the sip engine (10 is the most verbose output)
  • sip set debug on - sets debugging on of the sip engine
  • sip show users - shows configured users in sip.conf
  • sip reload - reloads the sip engine and rereads the sip.conf file
  • dialplan reload - reloads the dialplan enginer and rereads extensions.conf file
  • core show applications - shows all the applications which can be used in your dialplan
  • core show functions - shows all the functions which can be used while building a dialplan

There are many more commands, but I will not cover them all here obviously.

Final Words

By now you should have a basic fully working PBX system. What you need to do is to add more users/extensions, make sure you create a dial plan for them too and configure your phones/softphones and you can start making calls for free.

Final words

I do apologise if I made some mistakes or typos as I was writing everything out of my head, I did not have to test this actual setup. If you find any mistakes please let me know and I will correct that. Otherwise I hope it is going to be useful for somebody.