<div>Dear All,</div>
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<div>I am having this strange problem on my Asterisk 1.2.1. We have a web dialer that can register to the Asterisk box in Hong Kong, but another user using the same account can't register to the Asterisk box using the same web dialer. Below is an output of the sip debug logs. It seems that the digest is missing the username and password, but why? I have also have this call flow for the an IP Phone, but after a while, it will register to the Asterisk. One thing I don't understand is that I have registered successfully in Hong Kong and when the user tries in South Africa, it doesn't work. Please Help!
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<div>SIP Logs:</div>
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<div>From: XXXXXXXX <<a href="mailto:sip:XXXXXXXX@XXX.XXX.XX.XXX">sip:XXXXXXXX@XXX.XXX.XX.XXX</a>><br>To: XXXXXXXX <<a href="mailto:sip:XXXXXXXX@XXX.XXX.XX.XXX">sip:XXXXXXXX@XXX.XXX.XX.XXX</a>><br>Call-ID: <a href="mailto:1d3c5f0-6d7f-43cc33ac@XXX.XXX.XX.XXX">
1d3c5f0-6d7f-43cc33ac@XXX.XXX.XX.XXX</a><br>CSeq: 2 REGISTER<br>Contact: *<br>User-Agent: VaxSIP UserAgent/1.0<br>Expires: 0<br>Max-Forwards: 70<br>Content-Length: 0</div>
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<p><br>--- (11 headers 0 lines)---<br>Using latest REGISTER request as basis request<br>Sending to <a href="http://192.168.0.3">192.168.0.3</a> : 2232 (non-NAT)<br>Transmitting (NAT) to <a href="http://196.38.228.123:5060">
196.38.228.123:5060</a>:<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP <a href="http://192.168.0.3:2232">192.168.0.3:2232</a>;received=<a href="http://196.38.228.123">196.38.228.123</a><br>From: XXXXXXXX <<a href="mailto:sip:XXXXXXXX@XXX.XXX.XX.XXX">
sip:XXXXXXXX@XXX.XXX.XX.XXX</a>><br>To: XXXXXXXX <<a href="mailto:sip:XXXXXXXX@XXX.XXX.XX.XXX">sip:XXXXXXXX@XXX.XXX.XX.XXX</a>><br>Call-ID: <a href="mailto:1d3c5f0-6d7f-43cc33ac@XXX.XXX.XX.XXX">1d3c5f0-6d7f-43cc33ac@XXX.XXX.XX.XXX
</a><br>CSeq: 2 REGISTER<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Max-Forwards: 70<br>Contact: <<a href="mailto:sip:XXXXXXXX@XXX.XXX.XX.XXX">sip:XXXXXXXX@XXX.XXX.XX.XXX
</a>><br>Content-Length: 0</p>
<p><br>---<br>Transmitting (NAT) to <a href="http://196.38.228.123:5060">196.38.228.123:5060</a>:<br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP <a href="http://192.168.0.3:2232">192.168.0.3:2232</a>;received=<a href="http://196.38.228.123">
196.38.228.123</a><br>From: XXXXXXXX <<a href="mailto:sip:XXXXXXXX@XXX.XXX.XX.XXX">sip:XXXXXXXX@XXX.XXX.XX.XXX</a>><br>To: XXXXXXXX <<a href="mailto:sip:XXXXXXXX@XXX.XXX.XX.XXX">sip:XXXXXXXX@XXX.XXX.XX.XXX</a>>;tag=as63889026
<br>Call-ID: <a href="mailto:1d3c5f0-6d7f-43cc33ac@XXX.XXX.XX.XXX">1d3c5f0-6d7f-43cc33ac@XXX.XXX.XX.XXX</a><br>CSeq: 2 REGISTER<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br>Max-Forwards: 70<br>Contact: <<a href="mailto:sip:XXXXXXXX@XXX.XXX.XX.XXX">sip:XXXXXXXX@XXX.XXX.XX.XXX</a>><br>WWW-Authenticate: Digest realm="asterisk", nonce="4929aec7"<br>Content-Length: 0
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<p>Regards,</p>
<p>Kengie</p></div>