QUESTION Remote Extension Help with unusual PBX configuration

[Kaz Maynard]

Dear fellow PBXers those who are incredible and those who do it in a flash,

Firstly, I'm going to explain how this installation is setup at this particular client.

I am running IncrediblePBX 13-12.0 on a Supermicro 1u server. As with most supermicro servers it has two onboard ethernet ports. The telephone service provider/internet service provider brings triple play services into the premises via fiber into an ONT/modem, where the services are distributed as outlined:

ISP.........................................................................CLIENT
.....................10.251.0.135
PSTN---------MetaSwitch------- ...................................ONT4---GW:10.34.176.1---PBX eth1:10.34.176.28
.................................................|___FIBER___ONT__|
.................................................|............................|
.................................................|............................|__ ONT1---GW:192.168.0.1
INTERNET------ISP GW-------......................................................... |
....................................................................................................SWITCH-------PBX eth0:192.168.0.200
...........................................................................................................|
....................................................................................................Extensions

GW=Gateway
The ONT also has a WAN IP and behaves as a NAT Router only for ONT1 traffic.

I created static routes on eth1 so only packets destined for the metaswitch will travel that route, all other traffic is routed via eth0. The metaswitch is configured as a non registering sip trunk and sip traffic is perfect between the pbx and the metaswitch fabric. Call flow between extensions and PSTN via the metaswitch is perfect.

Here is my problem:
In order for rtp traffic to flow to the metaswitch, the IP of eth1 must be placed as the external IP in incrediblegui's asterisk sip settings and in the chan sip subsetting and nat must be turned off.

Thus the uri for communication between the PBX and the MetaSwitch is <dialedDID>@<eth1 IP> to <trunkDID>@<MetaSwitch IP> and everything works perfectly.
Now remote extentions require <dialedDID>@<wan IP>, but they will get <dialedDID>@<eth1 IP> as well and thus no audio is passed and the call will drop with a retransmission error.

Is there anyway for nat to be auto and external IP be the wan IP for remote extensions
and nat to be off and external IP be eth1 IP for just the Sip trunk?

I hope my diagram shows up properly and thanks in advance for any input or suggestions,

Kaz

PS: If you're ever trying to connect an asterisk 13 based pbx to a metaswitch do yourself a favor and configure chan_sip to use port 5060 and jsip to use 5061, it is reversed by default then create a sip trunk not a jsip.