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[codecomplete at free.fr]

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.


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[msc at freeswitch.org]

Just following up... did you get these questions ironed out?
-MC

On Wed, Mar 25, 2009 at 2:51 AM, Gilles <codecomplete@free.fr> wrote: