[asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available
barryf-lists at flanagan.ie
Sun Apr 27 07:34:45 CDT 2014
On 26 April 2014 23:32, James Cloos <cloos at jhcloos.com> wrote:
> 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.
...should do the job.
Hope this helps.
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