On 1/27/2011 12:53 PM, Matt White wrote:
>>> "Matthew Kitchin (public/usenet)" 01/27/11 1:42 PM >>>
>>There is no IP NAT. It is a private MPLS connection to Verizon. They
use public IP ranges in these setups, but there is no >>IP nat
involved. Their 172.x network and our 10.x network in this example
have direct non nat's access to each other.
Ok, yup, i see the ip is a 172...was thinking it was a 173.
But in you last post you say that you NAT port 5060 to 5080? So does
the router that sits between 172.x and your 10.81 only route, or does
it NAT? I would suspect its actually NATing a 172.x.x.x IP to the
10.83 side if your changing the port.
It is not NATing the IP. Their subnet has direct (routed) access to
ours. The router sits between these 2 subnets and in this case only
translates the port. Not the IP. It is a Verizon DS3, and we have 100
MPLS sites coming in through it. Also, in case it is relevant, I have to
put 'some' info on the NAT setting under servers in the sipx gui. I
don;t use STUN, so I change it to Specify an IP and I have to put
something in. It was trial and error 1.5 years ago when we started this.
I put in the IP for the Verizon gateway we send our calls. I know that
isn't correct, but I have to put some value in. I'm not sure if that is
relevant. The only other setting we change is 'Use public address for
call setup' under Gateway, ITSP account. We have to uncheck that box.
But your polycom endpoint is sending reinvites directly to
SipxBridge. And I've never seen this (and don't think this should
ever happen...correct me if I'm wrong).
The phone should send all sip signaling to the proxy...which then
sends it to sipxbridge. If you figure that out....you'll be all set.
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