Simple Configuration Asterisk Server PBX With SIP Client


Asterisk is a open source software based on Linux OS. Asterisk is an implementation of a PBX (private branch exchange) created by mark spencer of Digium.Inc in 1999.

Basically Asterisk is a voip server, asterisk has many features that are available in the PBX systems such as  voicemail, conference bridge, call queue and call detailed record. Users can create new functionality by writing dial plan scripts in Asterisk extension languages.

Asterisk supports a wide range of multimedia features such as Voice over IP protocol, using the protocol Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and H.323. Asterisk has a good roles as a registrar or as a gateway between VOIP and PSTN. 

This is a simple configuration between Asterisk PBX with SIP Client. In this case Voip (Voice Over Internet Protocol) Client that used is X-Lite SoftPhone. This softphone is free to use, and you can get it in the X-Lite site. If you want to use other free softphone for testing, you can read my references regarding free soft-phone.


1. Install Asteriks on Ubuntu with the following command 
apt-get install asterisk
2. Configure SIP file on /etc/asterisk/sip.conf. You can use vi text editor to edit sip.conf
vi  /etc/asterisk/sip.conf
[100]
type=friend
context=my-phones
secret=100
host=dynamic
[200]
type=friend
context=my-phones
secret=200
host=dynamic
[300]
type=friend
context=my-phones
secret=300
host=dynamic
Notice :
1. Sip.conf is used to add a user that will be registered on the PBX asterisk.
2. [100] = the name of user
3. Secret = user password
4. Type friend = to allow users can make calls and receive calls

4. Configure file extensions as below on /etc/asterisk/extensions.conf. 
vi /etc/asterisk/extensions.conf 
[my-phones]
exten => 100,1,Dial(SIP/100,20)
exten => 200,1,Dial(SIP/200,20)
exten => 300,1,Dial(SIP/300,20)
Notice :
Extensions.conf file is used to give extension to the user. The above examples are made extension 100 and destined for each user who has been mentioned in sip.conf. Next, go to the command line interface on the asterisk by giving the command :
asterisk -rvvv
 display screen as below :
Asterisk 1.6.2.5-0ubuntu1.4, Copyright (C) 1999 – 2009 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.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.
=========================================================================
== Parsing ‘/etc/asterisk/asterisk.conf’:   == Found
== Parsing ‘/etc/asterisk/extconfig.conf’:   == Found
Connected to Asterisk 1.6.2.5-0ubuntu1.4 currently running on xibo-server (pid = 10955)
Verbosity is at least 3
xibo-server*CLI>reload

Please see the picture below 

X-Lite Configuration
Notice:
User ID = users that has been created in sip.conf
Domain = IP Address Asterisk Server PBX
Password = Password user that has been created in sip.conf 

If you want to test it with Yealink IPPhone, you can get a setup guide on the official site of Yealink IP Phone.

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