Skip the whole NAT scenario.
Put up an asterisk box with two network interfaces. One interface
connects to the real world on your new IP address from your new ISP.
The other interface can be on the same subnet as the windows box that
you can't change. Set up a SIP trunk to your Windows box.
Nivin Kumar schrieb:
> Is there a tool that will allow me to automatically change sip headers
> in realtime?
Hi,
imho changing the SIP headers will not be sufficient, since
the "old" IP addresses are now private IP addresses (only in
your network, outside, there are still public, but pointing
not
- "Nivin Kumar" wrote:
>
Hello,
I'm in a bit of a fix. We have a particular Windows based softswitch which is
has its SIP and H323 ports hardcoded to listen on a particular IP address. The
problem is that the ISP is having major issues and we can no longer depend on
them for service. Th
Hello,
if the remote side (the public IP side) is capable to do
something like asterisk's nat=yes (in sip.conf), than
a mascerading router (like every cheap DSL router) would
do enough NAT do let SIP work.
If the remote side does not support that nat-hack (which
is not SIP standard), than you wil
Is there a tool that will allow me to automatically change sip headers in
realtime?
--- On Wed, 26/5/10, Motiejus Jakštys wrote:
From: Motiejus Jakštys
Subject: Re: [asterisk-users] Help with IP Routing
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Date: Wednesd
Discussion
Subject: Re: [asterisk-users] Help with IP Routing
Assume previous IP is LAN. Forwarding public IP ports to LAN is
straighforward. However, with SIP headers you will (don't know H323)
have to modify outgoing SIP headers: replace LAN ip with WAN ip.
For callers you have to substitut
Assume previous IP is LAN. Forwarding public IP ports to LAN is
straighforward. However, with SIP headers you will (don't know H323)
have to modify outgoing SIP headers: replace LAN ip with WAN ip.
For callers you have to substitute RTP destination IP
For callees you have to substitute RTP source I