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? thanks! -- _____________________________________________________________________ -- 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
