[Asterisk-Users] Newbie Question-Looking for Feedback

[woody+asterisk at solutionsfirst.com.au]

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> Christopher Raper
> Sent: Thursday, 8 January 2004 10:06
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Newbie Question-Looking for Feedback
> 
> Greetings all. I am new to the Asterisk world! Found it very 
> impressive so far!

Good.  And you are in Australia too!  For all the seppos who don't know
about Australia, it is just like Iowa except we have pet kangaroos instead
of pitchforks :-)

> In relation to the below.
> 
> I have worked with Alcatel PBX's for the last 3 years. 
> Alcatel OxE supports SIP and H323 as well.

Asterisk does both too, as well as IAX which works very well through NAT.

> As far as SIP goes I have also found the Xlite to be good for 
> soft phones. I am using one now.
> Check out www.xten.com Xlite is free and easy to use. I also 
> have been given a Pingtel SIP to play with.
> http://www.pingtel.com/ 

Many on the list use Xten, and I've heard of Pingtel, I'm not sure how well
it goes with Asterisk,
many use Grandstream, Snom, or Ci$co hardphones, and DIAX softphone.

> As far as H323 terminals go I have 
> not played with all that many, however the simple Microsoft 
> netmeeting works for testing purpose anyway.

http://openh323.org/ has many H323 apps for linux

> Now a question to all you experts out there, and this may 
> seem VERY stupid, but I have configured the sip phone and 
> have it logged in and can dial 500 to get to the sample 
> messages etc. However i cannot work out how to give the sip 
> termainal a number that can be dialled. I would assume that 
> it needs to be in the dialplan, so I have added it in via the 
> extensions.conf file, however I am sure that I have stuffed 
> the config somewhere. Can someone please point me in the 
> right direction. Would be much appreciated. Also, do i need 
> hardware to make a SIP to SIP call... eg. Compressors etc.

You also need to define the extension in /etc/asterisk/sip.conf

E.g.

[woody]
type=friend
insecure=yes
username=woody
secret=bogus
host=dynamic
defaultip=192.168.2.76

And in extensions.conf:

exten => 1976,1,Dial(SIP/woody,15,tr)

You might want to look at the wiki http://www.voip-info.org/wiki-Asterisk
which is a good place to find out how to do stuff with Asterisk.

cheers,
Woody