--- Repost: Sorry if you've already seen this message --- Hello,
We are using sipx 4.2.1 and have sipxbridge working fine for a couple of outbound trunks. Our SIPX server and phones are behind a NAT. I have been playing around with SIP URI dialing, and I am not sure how the media relaying/NAT busting is supposed to work. In my experimenting, I can get a call to connect, but I get no RTP. Upon looking further, it looks as if the outgoing RTP is being directed directly from the phone to the remote URI with no relay. (I would rather the RTP be relayed so that I don't have to open up my firewall for the phones). I am receiving no incoming RTP at all, which suggests to me that there is no NAT compensation being done. Anyway, I don't have any immediate need for URI dialing, but am just curious if this is supposed to work, or if there is any way to configure the behavior here. Thank You, Scott Richesson IT Manager Cincinnati Fan _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
