[Asterisk-Users] SIP redirect /New subject/

[Olle E. Johansson]

 > If a call comes to Server1 by SIP, is it possible to re-direct client to
> Server2. In another words, IAX2 part is (taken out), so client
> communicates with Server2 by SIP directly during that call. My primary
> motivation behind this is to save on resources.

How would you configure this? Always redirect some extensions or only if
something happens? Please explain a bit more.

I don't think it's possible today. Requires some additions.

/O