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 ([EMAIL PROTECTED]) 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
