[asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

[James Cloos]

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 <cloos at jhcloos.com>         OpenPGP: 0x997A9F17ED7DAEA6