[Asterisk-Users] How to route all incoming call to the defines context in extensions.conf

[jb at v2tel.com]

I'm testing the Asterisk in a pure sip configuration, presently testing it with
a number of sip phones, some registrations to a SER-server and with password
protection for outgoing calls to the SER-server.

I have a problem with incoming calls. When I get an incoming call, Asterisk
finds a peer in sip.conf and tries to route the call to that peer.

Apparently this happens because the address of the incoming call includes a
domain name with the same ip address as the ip address of the domain name of
the host entry in my [sipout] definition in sip.conf.

I would like to route all incoming calls to my extension.conf [sipin] heading,
even when a "peer is found".

If I delete the [sipout] definition in the sip.conf, I receive all incoming
calls in the way I want, but I cannot make outgoing calls. If I could include a
username and password in the dial command, I could do away with the [sipout],
but I have found no way to include this in the dial command.

I would appreciate any suggestions to solve the problem.

Thanks, Jon Bruel