[Asterisk-Users] setting up ---- newbie

[Greg Hill]

On Sun, 1 Feb 2004, Francisco Perez-Landaeta wrote:
> i am getting today my dev kit with fxo and fxs boards. i intend to do
> the following :
>
> 1) be able to route an incoming call from the pstn fxo port to an ip
> (answering with netmeeting or anyother sip softphone)
>
> 2) be able to call from netmeeting to my pstn fxo port to place calls.
>
> questions :
>
> how can i do this ? what are the commands for this simple setup ?

get asterisk (cvs or tarball). make. make install. make samples. Now
you'll have a "working" demo setup which you can modify to with with your
sip softphone. I don't know what you'll have to do to make the fxo and fxs
boards work; I haven't played with any yet.

Modifications you'll need to make:
- add a service definition in sip.conf so that your softphone can log in
to * using SIP. There are examples in the demo sip.conf file.
- edit extensions.conf. Write an extension which makes calls arriving on
the fxs port get dialed to your sip phone. Also write an extension which
lets you dial out the fxs port to the pstn. Again, there are examples in
the file.

check out the wiki, www.voip-info.org, for information about what goes
into sip.conf and extensions.conf.

> how can i place calls using a webbrowser (explorer, etc ?)

there are IAX and SIP applets around, but I can't recommend any (haven't
tried 'em)

> can i use messenger to call to call my pstn port ?

if it'll connect to *, then the rest of the magic happens when you write
the appropriate lines in extensions.conf.

> can i translate h323 to sip and viceversa ?

I think so.. I believe * accepts h323 connections as well as SIP, and it
would have to un-pack one protocol, possibly transcode to a different
codec, and then re-pack for transmission in the other protocol.

Greg