What are you using for a pstn gateway? The Mitel does send an sdp with the private address, dns address, or 0.0.0.0 in some occasion unless prohibited.
Here is an edited copy of a call with no transfer ...
Initial invite:
INVITE sip:blank@1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4c88f7a994f2db72f130519551c5fedc.0
From: WIRELESS CALLER <sip:blank@blank:5060>;tag=A7C2.9B55
To: <sip:blank@blank:7061>
Call-ID: 0026.9E9B.B4F2.4DD2.A7C2.1B5A
CSeq: 8738 INVITE
Max-Forwards: 69
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4DD2.A7C2.9B558738.A48B6018.00001B95.zCp3O4Kz4K+KbJ34qyDi3Q__
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4DD2.A7C2.9B558738.-aN6AgMoWp89KmEvOe6hxw__
Contact: <sip:e4coEs7GA1Rbz1_ILEA3EVBH-hVHznloebVs7mglKAoOk-M9uL0tNDEQTg1c2kZ7r@1.2.3.4>
Session-Expires: 1800;refresher=uac
Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE
Supported: timer,100rel
Content-Type: application/sdp
Content-Length: 252
Record-Route: <sip:7c5669fe4b5976c9@1.2.3.4;lr>
v=0
o=BroadSoft 25576 25576 IN IP4 1.2.3.4 (Public SDP Owner)
s=M6 Call
c=IN IP4 1.2.3.4
t=0 0
m=audio 58238 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:40
200 OK:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4c88f7a994f2db72f130519551c5fedc.0,SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4DD2.A7C2.9B558738.A48B6018.00001B95.zCp3O4Kz4K+KbJ34qyDi3Q__,SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4DD2.A7C2.9B558738.-aN6AgMoWp89KmEvOe6hxw__
Record-Route: <sip:7c5669fe4b5976c9@1.2.3.4;lr>
From: WIRELESS CALLER <sip:blank@blank:5060>;tag=A7C2.9B55
To: <sip:blank@blank:7061>;tag=0_2334818960-92952406
Call-ID: 0026.9E9B.B4F2.4DD2.A7C2.1B5A
CSeq: 8738 INVITE
Supported: timer
Contact: <sip:blank@1.1.1.1:5060;transport=udp>
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
Content-Type: application/sdp
User-Agent: Mitel-3300-ICP 10.0.1.21
Content-Length: 162
v=0
o=- 9119 9119 IN IP4 1.2.3.4 (Private SDP Owner Proxies' 200 OK Shows Public)
s=-
c=IN IP4 1.2.3.40
t=0 0
m=audio 50278 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
I have found thru a lot of SIP work it is very wise to use an Ingate proxy as it helps to fix some of the signalling issues that you see between pbxs and soft switches. It also does the nat translation or routing which some firewalls even ones claiming to be sip aware fail on.