[Asterisk-Users] Question: handling fully-qualified SIP dial requests

[John Todd]

I have Cisco 79xx phones on the desktop here, which are capable of 
alpha input through the numeric keypad.  I'd like to place calls to 
fully-qualified SIP addresses (user at foo.edu) but it seems that 
Asterisk is somehow stripping the @foo.edu part of the request off 
when the entry hits my dialplan, and only the username is being 
matched.  I'd like to just match on any string that contains "@", and 
then hand those calls over to a very simple Dial statement.

Does anyone have experience with passing fully-qualified SIP 
addresses through the Asterisk dialplan that can give me some hints?

Note: Yes, I want Asterisk to be in the call flow and I do not want 
to make the calls "directly" from the SIP device to the other end due 
to firewall/NAT/access control issues.

JT