[SR-Users] Route calls from Kamailio to Asterisk behind NAT

[Benjamin Fitzgerald]

Yes, it requires a very particular configuration. With asterisk you in
sip.conf you need

externaddr=you.public.ip

This is used in SIP registrations by Asterisk. You should also be
comfortable with debugging the SIP traffic to make sure all IP's are set
properly. For asterisk:

>sip set debug on

Look for REGISTER messages.

Another amazing tool for this is sipgrep (
https://github.com/sipcapture/sipgrep). It's like ngrep but for SIP.



Benjamin Fitzgerald
LETS Corporation
(925) 566-5600 ext. 25
ben at letscorp.us




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On Fri, Aug 11, 2017 at 3:29 AM, <vs at cell.tel> wrote:

> Hi!
>
> Could anyone advice me or provide an example of Kamailio config to solve
> this case:
>
> Current Kamailio+Asterisk schema:
> (7XXX Users) <-> Asterisk <-> NAT <-> Internet <-> Kamailio <-> (4XXX
> Users)
>
>
> 7XXX Users registers on Asterisk server, and 4XXX Users registers on
> Kamailio.
> In Addition, Asterisk registers on Kamailio with user 4444, and it is
> possible to make calls from 7XXX Users (Asterisk) to 4XXX Users (Kamailio)
> through 4444 "trunk".
> Also, while Kamailio users (4XXX) call to 4444 number, call successfully
> goes to Asterisk.
>
>
> How its possible to route all 7XXX destined calls from Kamailio users to
> Asterisk, while Asterisk is behind NAT???
>
> Config examples, i found, were writen only for situation with fixed
> asterisk IP and PORT, without nat ( asterisk.bindip = "" ; asterisk.bindport
> = "")
>
>
>
> Thanks in advance
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
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