[OpenSIPS-Users] inbound "failoiver"

[Dmitri G.]

Hello,

I would like to implement some kind of failover for my Asterisk, let me
describe how I would like to see it.


I registered 2 sip users with my Kamailio, 1020 at domain.com and
1030 at domain.com.
I have added aliases to 1020 (aliases from 1021 at domain com to
1029 at domain.com).

Right now calls to 1020-1029 goes well to 1020, it works fine.

But I would like to do the following:

If 1020 isn't registered with Kamailio (let's say if registration for 1020
is down in Kamailio, so AOR not found for 1020), it is possible to route
calls to 1030?
So route calls to 1030 only when registration for 1020 isn't active in
Kamailio.

I have tried manipulating with faillure_route, but without any luck.


I hope it is clear what I would like to do :)


Any help would be much appreciated.


Kind regards,

Dmitri
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