You can also leave the server untouched and add a route on your (SIP/RTP) 
default-gateway (192.168.1.6):
 
ip route 172.16.30.0 255.255.255.0 192.168.1.1.

All your SIP/RTP traffic will flow as usuall. All the Web/Gui traffic will 
flow via 192.168.1.6 to 192.168.1.1
 
If you also send ICMP-redirect on the D-GW (cisco's do it by default) and 
accept them on the server (small security risk, don't know defaults) 
then the Web/Gui traffic will flow directly to 192.168.1.1 after the first 
packet.

http://www.cisco.com/en/US/tech/tk365/technologies_tech_note09186a0080094702.shtml
 


This solution has the advantage that the SipX box can be 
replaced/rebuild/upgraded etc without having to think about the routing.

Paul

"[email protected]" <[email protected]> wrote:
> 
> > Why are there 2 Gateways in the Network of your sipx? Which nets are 
they
> > connected to?
> 
> I have several gateways on the network, each for it's own traffic to
> prevent packets flowing on to network segments they don't belong on.
> It's also a security measure of course.
> 
> > If there is only one specific IP Network connected to the 192.168.1.1 
then
> > you can add a specific route only for this network and the default 
gateway
> > for all other networks is still the 192.168.1.6.
> 
> So, you mean for example, 
> route add -net 172.16.30.0 255.255.255.0 192.168.1.1
> This would allow traffic destined back to 192.168.1.1 to flow back 
> out to that gateway without affecting the rest of the traffic? 
>  
> > This should not cause any effects as long as you have only unique
> > networks/IP-addresses involved.
> 
> In this case, I have a sipx server at 192.168.1.x and it's gateway 
> is 192.168.1.6 for SIP/RTP connections only.
> I have web users coming from 172.16.30.x who need access to the 
> dashboard on sipx so when they connect, the packets don't flow back 
> to them since the servers gateway is 192.168.1.1.
> 
> If there isn't a problem adding a second route so that web (8443) 
> connections can flow back to the 172.16.30.x network without messing
> up SIP/RTP traffic, then that seems to be the solution. 
> 
> This is what I am looking to confirm so that I don't do something lame.
> 
> Mike
> 
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