On Fri, 26 Feb 2010, Alejandro Recarey wrote: > I'm trying to get my asterisk server to reinvite. I have two asterisk > servers with public IP's. My users (behind NAT) register on one server > (I'll call it server 1), and for some calls they are transfered over > to the other server (server 2), because that server has the E1's. > > I want server 1 to be in the signaling path for billing reasons, but > handling the media stream is killing my capacity, and it should not be > necessary as server 2 also has a public IP address. > > I have tried playing around with the "canreinvite" options in sip.conf > but the problem is I cannot tell if asterisk is reinviting the call or > not. > > How can I figure out where the media stream is going?
Running iftop on the box in a terminal window makes it relatively easy to see what's going where. There are lots of things in asterisk that'll stop this working though - mostly detailled here: http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite Good luck! Gordon -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
