[Asterisk-Users] SIP routing over IAX2

[Bart J. Smit]

Hi All,

I have two Asterisks, one on the LAN that handles the internal calls
with a PSTN interface and one on the DMZ with a public interface. I
would like to route SIP calls from the internal users to the Internet
over IAX2 to the DMZ and onwards.

All users have soft phones so they would enter sip:someuser at somevoip.org
to get a connection. I would like to avoid having number prefixes to
dial external SIP phones.

Can Asterisk do this? I am relatively new to Asterisk. I guess I'm after
something like an email smarthost feature for SIP.

I have googled and checked some of the getting started pages but all
dial plans deal with number prefixes to route calls. I want to route
calls starting with 'sip:' as a prefix.

Thanks,

Bart...