Hello, I have a couple of questions related to having SIP users connecting from the Net to a Freeswitch server through NAT routers on both ends: 1. How must I configure routers on both ends? I understand that I need to route incoming TCP/UDP 5080 into the Freeswitch server, but what about the other router? I guess I also need to route this port to let the SIP phone ring, but what about data (RTP/RTCP)? 2. The Freeswitch server is connected to the POTS with either an OpenVox PCI card or a Linksys 3102 box: When a call is made between the POTS and a remote SIP phone (ie. out there on the Net, not on the same LAN as the Freeswitch server), is there a way for data to flow directly from the POTS to the remote SIP client instead of through the Freeswitch server? Thank you.