[Asterisk-Users] Asterisk as SIP Proxy

[ranga]

Hi,

I started evaluating asterisk about a week back. I was trying to configure
asterisk as SIP proxy.
This is the setup that I have now.

I have one linux box running asterisk ( say 192.168.68.15 ) and second box
running partysip (say 192.168.68.6).

I am using SJPhone from windows boxes.

I registered one user to partysip -john at 192.168.68.6
and another user to asterisk -mark at 192.168.68.15

I had this entry in extensions.conf

exten => john,1,Dial(SIP/john at 192.168.68.6)

It worked well. I was able to call john.

Now this is what I wanted to achieve.
    The other sip server ( here partysip) may have many users registered. It
is not possible to make every user's entry into extensions.conf. Instead,
any mechanism where I can replace 192.168.68.6 with a variable that
represents the 'To' domain will be a great. In simple, I am looking for a
line in extensions.conf that looks like following.
exten => _proxy-.,1,Dial(SIP/${EXTEN:6} @ ${DOMAIN})

*variable DOMAIN is my assumption.

This is the possible soln that I can think off. This, I guess, needs little
hack into chan_sip.c. Is there any other way that simplifies this task?

thanks
-Ranga