[Freeswitch-users] [Remote SIP client] Couple of questions

[Gilles]

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.