Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki?
Blake _______________________________________________ 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
