[Asterisk-Users] SIP NAT Polycom

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Hi,
have a setup which should not be unknown to others;
Asterisk behind wall doing NAT, and out in the wild world behind linksys
router a Polycom phone. The Polycom phone is on DMZ. It should register
with my server.
sip conf:
        [4031]			
        type=friend
        context=main
        callerid="HJEMME" <4031>
        secret=4031
        nat=yes
        canreinvite=yes
        qualify=yes
        host=dynamic
        dtmfmode=rfc2833 
        username=4031
        mailbox=4031 at main
        disallow=all
        allow=ulaw 
        allow=gsm
        progressinband=no

I can dial and the phone rings,caller ID comes up, both ways this is
working.
Now answering, gives no sound, or just a fraction of a second of sound.
Voicemail, greetings etc appears to use ports above 10000, these seems
to pass ok.
Debugging RTP, which I read is where the audio is passed,and which has
been opened on server side gateway to ports 10000-20000, showed packets
to the Polycom address on ports 29xxx, whereas I opened rtp ports upto
30000.

No change, below the initiating sequence of a call being answered.

           -- SIP/4031-2264 answered SIP/4030-1c2e
            -- Attempting native bridge of SIP/4030-1c2e and
        SIP/4031-2264
        Got RTP packet from 10.1.0.51:10080 (type 0, seq 39101, ts
        368481408, len 160)
        Sent RTP packet to xx.xx.x.xxx:29256 (type 0, seq 35268, ts -32,
        len 160)
        Got RTP packet from 10.1.0.51:10080 (type 0, seq 39102, ts
        368481568, len 160)
        Sent RTP packet to xx.xx.x.xxx:29256 (type 0, seq 35269, ts 128,
        len 160)
        Got RTP packet from xx.xx.x.xxx:29258 (type 0, seq 7541, ts
        1008244723, len 160)
        Sent RTP packet to 10.1.0.51:10080 (type 0, seq 19971, ts -80,
        len 160)
        

I am a bit confused about the settings of RTP on the Polycom 600, could
be something there.

Anyone that could get me on track?

regards
Frank