In 1.2 you can use rtp debug in the CLI On Thu, Feb 25, 2010 at 8:27 PM, Alejandro Recarey <[email protected]> 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? > > 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 >
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