View source for Asterisk

[[Category:Server]]
=Asterisk Configration=
==Summary==
Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny. 

This wiki will show you how to configure a simple in house network enabling us to use a SIP softphone to talk to another SIP softphone on your LAN. This is nothing major, but will get you familiar with Asterisk configuration and its pretty cool. 

==What You Need==
Of course, youll need [http://www.asterisk.org Asterisk], a SIP softphone and at least two machines. Recommendations for SIP phones are [http://www.wirlab.net/kphone/ kphone] and [http://www.xten.net/index.php?menu=X-Series x-lite].

==Configuration==
Assuming your asterisk server is up and running we will only need to edit two files sip.conf and extensions.conf. 
Change to your asterisk configuration directory (should be <code>/etc/asterisk</code>). Edit sip.conf and place the following in:
<pre>
[me1]
type=friend
username=me1
secret=password
host=dynamic
context=house

[me2]
type=friend
username=me2
secret=password
host=dynamic
context=house
</pre>

This creates our two SIP users <code>me1</code> and <code>me2</code> with a password of <code>password</code> in the <code>house</code> context. We will be defining the context next.

Edit extensions.conf with the following:
<pre>
[house]
exten => 100,1,Dial(SIP/me1)

exten => 101,1,Dial(SIP/me2)
</pre>

This creates that context <code>house</code> and assigns extension 100 to the SIP user <code>me1</code>, and extension 101 to the SIP user <code>me2</code>. Now all thats left is to see if it works.

==Asterisk Console And Softphones==
Now lets get Asterisk going. To do this enter
<pre>
# asterisk -vvvvvvc
</pre>
This will give us the Asterisk CLI with verbose output. If Asterisk is already running you'll need to use
<pre>
# asterisk -r
</pre>
Now fire up your SIP clients and set them up with the information in the sip.conf. Switch back to your Asterisk CLI and you should see 
<pre>
 Registered SIP 'me1' at 192.168.0.142 port 5061 expires 60
</pre>

Now you should be able to dial <code>101</code> from <code>me1</code> and talk to me2. 

==Common Problems==
If you recieve a 404 Not Found error check your extensions.conf and the number you dialed.

==Coming Up==
*Voicemail
*Connecting To The Outside
*Music On Hold