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Hello,<BR>
I tried to send the SIP call to AS5350 from a IP PBX but got one way audio.<BR>
The scenario is SIP phone ->NAT->IP PBX -> AS5350 ->T1 circuit.<BR>
When the call reached AS5350, it came with private IP address and sent the RTP back to it instead of public IP's.<BR>
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.<BR>
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======Debug output=======<BR>
(x.y.z.a=AS5350 x.y.z.b=IP PBX)<BR>
<P sizset="75" sizcache="5">Apr 6 14:15:23.866: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<BR>Received: <BR>INVITE sip:8411647@x.y.z.a SIP/2.0<BR>Call-ID: <A class=jive-link-external-small href="mailto:6a34022c7c7a0d2c6234438c553eceb2@172.16.1.179">6a34022c7c7a0d2c6234438c553eceb2@172.16.1.179</A><BR>CSeq: 2 INVITE<BR>From: <sip:test01@x.y.z.b>;tag=e002bb19cef9ffdee832<BR>To: <sip:8411647@x.y.z.a><BR>Via: SIP/2.0/UDP x.y.z.b:5061;branch=z9hG4bk-e002bb19cef9ffdee832<BR>Max-Forwards: 70<BR>Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE<BR>Contact: <sip:ippbx@x.y.z.b:5061><BR>Supported: replaces<BR>Content-Type: application/sdp<BR>Content-Length: 339</P>
v=0<BR>o=CMI-SIPUA 63439 0 IN IP4 172.16.1.179<BR>s=SIP CALL<BR><FONT color=#ff0000>c=IN IP4 172.16.1.179</FONT><BR>t=0 0<BR>m=audio 60000 RTP/AVP 0 8 4 18 23 22 2 21 101<BR>a=rtpmap:23 G726-16/8000<BR>a=rtpmap:22 G726-24/8000<BR>a=rtpmap:2 G726-32/8000<BR>a=rtpmap:21 G726-40/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-15<BR>a=fmtp:18 annexb=no<BR>a=rtcp:60001<BR>a=sendrecv<BR>
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Stream type : voice-only<BR> Media line : 1<BR> State : STREAM_ADDING (2)<BR> Callid : -1<BR> Negotiated Codec : g711ulaw, bytes :160<BR> Negotiated DTMF relay : inband-voice<BR> Negotiated NTE payload : 0<BR> Negotiated CN payload : 0<BR> Media Srce Addr/Port : x.y.z.a<BR> <FONT color=#ff0000>Media Dest Addr/Port : 172.16.1.179:44042</FONT><BR>
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