[cisco-voip] AS5300 Media address

[David Eco]

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.

Is there any way to set the media address as the IP's in FROM which means sending the media back to the origination, IP PBX? 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 at x.y.z.a SIP/2.0
Call-ID: 6a34022c7c7a0d2c6234438c553eceb2 at 172.16.1.179
CSeq: 2 INVITE
From: <sip:test01 at x.y.z.b>;tag=e002bb19cef9ffdee832
To: <sip:8411647 at 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 at 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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