And related thereto: What needs to be done on kama and ast to ensure that all incoming calls which route through a given kama box always matches a sip.conf [section] based on the socket(7)'s remote address, w/o any consideration of the INVITE's sip headers or body?
I tried a several variations, but nothing quite worked. -JimC -- James Cloos <[email protected]> OpenPGP: 0x997A9F17ED7DAEA6 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
