asterisk users - Sep 2004 - sound dropouts during SIP re-register

hi

I keep getting sound dropouts during SIP re-registration, and I can't  
find a remedy for it. I use SIP friends from MySQL. Below is SIP debug  
output for the re-registration

Thanks in advance

roy

-------

*CLI> sip debug ip 80.202.161.221
SIP Debugging Enabled for IP: 80.202.161.221
*CLI>

Sip read:
REGISTER sip:sipgw1.briiz.no SIP/2.0
From: sip:1001001@sipgw1.briiz.no;tag=Ypw9-mgYqF
To: sip:1001001@sipgw1.briiz.no
Call-ID: ApyiV0-ttr029@sipgw1.briiz.no
CSeq: 149 REGISTER
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK6E09-lHoEr00V
Contact: sip:1001001@10.0.0.3:5060
Max-Forwards: 70
Authorization: Digest  
username="1001001",realm="asterisk",uri="sip: 
sipgw1.briiz.no",response="4eebb62b14b2e8f2ca9d7ac6953a2bdf",nonce="4743
6d14"
User-Agent: 100/000003
Expires: 10
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 10.0.0.3 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  
10.0.0.3:5060;branch=z9hG4bK6E09-lHoEr00V;received=80.202.161.221; 
rport=49197
From: sip:1001001@sipgw1.briiz.no;tag=Ypw9-mgYqF
To: sip:1001001@sipgw1.briiz.no;tag=as5767c396
Call-ID: ApyiV0-ttr029@sipgw1.briiz.no
CSeq: 149 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1001001@213.160.242.5>
Content-Length: 0


  to 80.202.161.221:49197
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP  
10.0.0.3:5060;branch=z9hG4bK6E09-lHoEr00V;received=80.202.161.221; 
rport=49197
From: sip:1001001@sipgw1.briiz.no;tag=Ypw9-mgYqF
To: sip:1001001@sipgw1.briiz.no;tag=as5767c396
Call-ID: ApyiV0-ttr029@sipgw1.briiz.no
CSeq: 149 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1001001@213.160.242.5>
WWW-Authenticate: Digest realm="asterisk", nonce="706043b7"
Content-Length: 0


  to 80.202.161.221:49197
Scheduling destruction of call 'ApyiV0-ttr029@sipgw1.briiz.no' in 15000
ms


Sip read:
REGISTER sip:sipgw1.briiz.no SIP/2.0
From: sip:1001001@sipgw1.briiz.no;tag=Ypw9-mgYqF
To: sip:1001001@sipgw1.briiz.no
Call-ID: ApyiV0-ttr029@sipgw1.briiz.no
CSeq: 150 REGISTER
Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK6E09-tYypT9Bj0
Contact: sip:1001001@10.0.0.3:5060
Max-Forwards: 70
User-Agent: 100/000003
Expires: 10
Authorization: Digest  
username="1001001",realm="asterisk",uri="sip: 
sipgw1.briiz.no",response="ba23a26d03b3fcc888dce81fad859543",nonce="7060
43b7"
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 10.0.0.3 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  
10.0.0.3:5060;branch=z9hG4bK6E09-tYypT9Bj0;received=80.202.161.221; 
rport=49197
From: sip:1001001@sipgw1.briiz.no;tag=Ypw9-mgYqF
To: sip:1001001@sipgw1.briiz.no;tag=as5767c396
Call-ID: ApyiV0-ttr029@sipgw1.briiz.no
CSeq: 150 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1001001@213.160.242.5>
Content-Length: 0


  to 80.202.161.221:49197
     -- Registered SIP '1001001' at 80.202.161.221 port 49197 expires 10
     -- Saved useragent "100/000003" for peer 1001001
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP  
10.0.0.3:5060;branch=z9hG4bK6E09-tYypT9Bj0;received=80.202.161.221; 
rport=49197
From: sip:1001001@sipgw1.briiz.no;tag=Ypw9-mgYqF
To: sip:1001001@sipgw1.briiz.no;tag=as5767c396
Call-ID: ApyiV0-ttr029@sipgw1.briiz.no
CSeq: 150 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 10
Contact: <sip:1001001@10.0.0.3:5060>;expires=10
Date: Wed, 29 Sep 2004 09:40:40 GMT
Content-Length: 0


  to 80.202.161.221:49197
Scheduling destruction of call 'ApyiV0-ttr029@sipgw1.briiz.no' in 15000
ms