[asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available
cloos at jhcloos.com
Sat Apr 26 17:32:23 CDT 2014
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.
James Cloos <cloos at jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6
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