Help Article: Configuring X-Lite for Asterisk

This tutorial is not a comprehensive review of X-Lite. The purpose of this post is to set up a soft phone for evaluation. Soft phones can save money over the purchase of SIP handsets, but I strongly encourage you to test the soft phones before making a purchase decision. Soft phone quality varies widely depending on network conditions, codecs and protocol. This tutorial will help you evaluate the feasibility of soft phones with Asterisk. Before we start with X-Lite lets set up SIP and dial plan.

1) Change directory to zaptel source directory.
example:
[matt@localhost ~]$ cd /etc/asterisk

2) I'm using nano to edit the file, but pico, vi, emacs, or any text editor will do.
example:
[matt@localhost asterisk]$ nano sip.conf

[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 192.168.1.x ; x = Asterisk server IP address
allow = ulaw ; Allow all codecs
context = bogon-calls ; Send SIP callers that we don't know about here

[9250]
type=friend
username=9250
secret=password
host=dynamic
context=from-sip
mailbox=9250
nat=no
canreinvite=no

[9251]
type=friend
username=9251
secret=password
host=dynamic
context=from-sip
mailbox=9251
nat=no
canreinvite=no

3) Okay, we have the configuration for two clients on the SIP server. Now we have too make two extensions. These extensions will not include voicemail.
example:
[matt@localhost asterisk]$ nano extensions.conf

[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[bogon-calls]

[from-sip]

exten => 9250,1,Dial(SIP/9250,20)
exten => 9250,2,Hangup

exten => 9251,1,Dial(SIP/9251,20)
exten => 9251,2,Hangup

4) Now we need to download X-Lite and install on our Windows PC. Download the soft phone from http://www.xten.net/index.php?menu=download. Run the install. Open X-Lite click on the menu button.



Click on 'System Settings'. Then choose the 'SIP proxy' option. Click on default and continue to proceed with filling out the SIP client configuration. When you are done t should look like this:



5) Now you are ready to login and test some calls.

To Be Continued....Next Week

Note: My name is Matt Birkland, I work as a VoIP Engineer for VoiceIP Solutions an Asterisk Provider in the Seattle area. Every Week I will be submitting a one page Asterisk/VoIP tip of the week on the blog. Next week we will build on this subject by reviewing codes and at some point we'll move on to common network issues or IAX tunnels... haven't decided yet!