NAT and SIP

KruglerD

So I have an old PBX system, that does work with SIP in a fairly rudimentary method, but it does work. We were ready to switch to all sip (We have been using a POTS gateway up to this point) The system during placing the phone call on hold to switch to it's either automated attendant or voicemail, send a packet, which contains the local ip address instead of the public IP, which basically ends up cutting off audio because of the way our provider tries to limit latency by sending the RTP packet directly from the source.

Below is the packet that I think is messing up the system. I've x'ed out the personal info, but there are actual numbers where the "X"'s are, Just not pertinent to the conversation.

2022-02-11 22:56:11 +0000 : 75.XXX.XXX.217:1480
INVITE sip:+17152XXXXX2@23.XXX.XXX.86:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.XXX:9115;rport
Max-Forwards: 70
From: <sip:+17632XXXXX3@fl.gg>;tag=7102e573c9
To: "DAVID KRUGLER" <sip:+17152XXXXX2@fl.gg>;tag=gK0c28b70f
Call-ID: 571214035_40936028@23.XXX.XXX.86
CSeq: 2 INVITE
Route: <sip:34.226.36.32;lr>
Route: <sip:34.211.68.10;lr>
Contact: <sip:192.168.0.XXX:9115>
User-Agent: MSFTRP 2.0
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 215
v=0
o=MSRP 0 48 IN IP4 XXX
s=-
c=IN IP4 192.168.0.XXX
t=0 0
m=audio 41050 RTP/AVP 0 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

it's my opinion that the packet Contact: <sip:192.168.0.XXX:9115> that is being sent, the audio provider looses communication and nothing further is received. I have been looking at TURN, and siproxd trying to see if there was a way to globally substitute 192.168.0.XXX with 75.XXX.XXX.217 but nothing is jumping at me.
Can anyone recommend a solution or a how to article to this? Because the PBX is running WinXP, I'm trying to keep it protected as much as possible, so DMZ and other solutions are kind of out of the picture.

Thanks in advance.