Sorry, I have accidentally sent this message to opensips mailing list, I am sending it now to the right place 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(a)domain.com and 1030(a)domain.com. I have added aliases to 1020 (aliases from 1021@domain com to 1029(a)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