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

Barry Flanagan 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.
>
>

Something like:

[peer_inbound]
context=peercontext
type=peer
host=192.168.1.1


...should do the job.


Hope this helps.

-Barry Flanagan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140427/771aff1e/attachment.html>


More information about the asterisk-users mailing list