[Asterisk-Users] external SIP calls newbie question

[Chris Stenton]

I have  configured a basic * box which allows external sip calls in. This works correctly if someone calls say 23 at foo.bar and thus gets put through to extension 23. However, I have not been able to figure out or find the documentation on  how to direct sip calls made to just foo.bar to be sent to a default extension. In the sip debug it says its looking for the IP address of the caller in the default context which of course it cant find and rejects the call. Is there a way to force the extension?


Chris