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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
