Sorry, I have accidentally sent this message to opensips mailing list, I am sending it now to the right place<br><br><br><br>Hello,<br><br>I would like to implement some kind of failover for my Asterisk, let me describe how I would like to see it.<br>
<br><br>I registered 2 sip users with my Kamailio, <a href="mailto:1020@domain.com" target="_blank">1020@domain.com</a> and <a href="mailto:1030@domain.com" target="_blank">1030@domain.com</a>.<br>
I have added aliases to 1020 (aliases from 1021@domain com to <a href="mailto:1029@domain.com" target="_blank">1029@domain.com</a>).<br><br>Right now calls to 1020-1029 goes well to 1020, it works fine.<br><br>But I would like to do the following:<br>
<br>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?<br>So route calls to 1030 only when registration for 1020 isn't active in Kamailio.<br>
<br>I have tried manipulating with faillure_route, but without any luck.<br><br><br>I hope it is clear what I would like to do :)<br><br><br>Any help would be much appreciated.<br><br><br>Kind regards,<br><font color="#888888"><br>
Dmitri </font><br>