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As5350 media address

Hello,

I tried to send the SIP call to AS5350 from a IP PBX but got one way audio.

The scenario is SIP phone ->NAT->IP PBX -> AS5350 ->T1 circuit.

When the call reached AS5350, it came with private IP address and sent the RTP back to it instead of public IP's.

My question is how to set the media address as the IP's in FROM? Thank you.

======Debug output=======

(x.y.z.a=AS5350   x.y.z.b=IP PBX)

Apr  6 14:15:23.866: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:8411647@x.y.z.a SIP/2.0
Call-ID: 6a34022c7c7a0d2c6234438c553eceb2@172.16.1.179
CSeq: 2 INVITE
From: <sip:test01@x.y.z.b>;tag=e002bb19cef9ffdee832
To: <sip:8411647@x.y.z.a>
Via: SIP/2.0/UDP x.y.z.b:5061;branch=z9hG4bk-e002bb19cef9ffdee832
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Contact: <sip:ippbx@x.y.z.b:5061>
Supported: replaces
Content-Type: application/sdp
Content-Length: 339

v=0
o=CMI-SIPUA 63439 0 IN IP4 172.16.1.179
s=SIP CALL
c=IN IP4 172.16.1.179
t=0 0
m=audio 60000 RTP/AVP 0 8 4 18 23 22 2 21 101
a=rtpmap:23 G726-16/8000
a=rtpmap:22 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:21 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=rtcp:60001
a=sendrecv

          Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Callid                 : -1
          Negotiated Codec       : g711ulaw, bytes :160
          Negotiated DTMF relay  : inband-voice
          Negotiated NTE payload : 0
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : x.y.z.a
          Media Dest Addr/Port   : 172.16.1.179:44042

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