<br><br>
<div><span class="gmail_quote">On 10/3/05, <b class="gmail_sendername">Morten Isaksen</b> <<a href="mailto:misaksen@gmail.com">misaksen@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><br>
<div><span class="q"><span class="gmail_quote">On 10/3/05, <b class="gmail_sendername">Olle E. Johansson</b> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:oej@edvina.net" target="_blank">oej@edvina.net
</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">> Does anyone know what "stale nonce" is?<br>I've answered this question many times, so you should be able to find
<br>the answer...<br><br>A stale nonce is when a device tries to re-authenticate with a nonce<br>that is no longer valid. We are telling them that the nonce they used is<br>invalid, and re-issue a new challenge and a fresh nonce. It's just an
<br>informative message, that I propably should move away to a debug level<br>of some kind.</blockquote>
<div> </div>
<div> </div></span>
<div>I get this error when I use a Audiocodes MP-124 against Asterisk 1.2beta1 and asterisk refuses the call. When I use CVS-D2005.02.12.14.37.11-04/13/05-16:14:03 it works fine.</div>
<div> </div>
<div>I do not have access to the debug and log file now, but I will send them tomorrow.</div>
<div> </div></div></blockquote></div>
<div> </div>
<div>Here is the output from sip debug. I hope someone can explain what is wrong.</div>
<div><br>
<p><-- SIP read from <a href="http://10.131.2.1:5060">10.131.2.1:5060</a>:<br>INVITE <a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://10.131.2.1">10.131.2.1
</a>;branch=z9hG4bKaciipncbQ<br>Max-Forwards: 70<br>From: <<a href="mailto:sip:070001@10.131.0.1">sip:070001@10.131.0.1</a>>;tag=1c1850211233<br>To: <<a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone>
<br>Call-ID: <a href="mailto:195411465Zwlj@10.131.2.1">195411465Zwlj@10.131.2.1</a><br>CSeq: 1 INVITE<br>Contact: <<a href="mailto:sip:070001@10.131.2.1">sip:070001@10.131.2.1</a>><br>Supported: em,100rel,timer,replaces,path
<br>Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE<br>User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006<br>Content-Type: application/sdp<br>Content-Length: 242</p>
<p>v=0<br>o=AudiocodesGW 644554 101011 IN IP4 <a href="http://10.131.2.1">10.131.2.1</a><br>s=Phone-Call<br>c=IN IP4 <a href="http://10.131.2.1">10.131.2.1</a><br>t=0 0<br>m=audio 6070 RTP/AVP 8 0 96<br>a=rtpmap:8 pcma/8000
<br>a=rtpmap:0 pcmu/8000<br>a=rtpmap:96 telephone-event/8000<br>a=fmtp:96 0-15<br>a=ptime:20<br>a=sendrecv</p>
<p>--- (13 headers 12 lines)---<br>Using INVITE request as basis request - <a href="mailto:195411465Zwlj@10.131.2.1">195411465Zwlj@10.131.2.1</a><br>Sending to <a href="http://10.131.2.1">10.131.2.1</a> : 5060 (non-NAT)<br>
Reliably Transmitting (no NAT) to <a href="http://10.131.2.1:5060">10.131.2.1:5060</a>:<br>SIP/2.0 407 Proxy Authentication Required<br>Via: SIP/2.0/UDP <a href="http://10.131.2.1">10.131.2.1</a>;branch=z9hG4bKaciipncbQ<br>
From: <<a href="mailto:sip:070001@10.131.0.1">sip:070001@10.131.0.1</a>>;tag=1c1850211233<br>To: <<a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone>;tag=as6a339401<br>Call-ID: <a href="mailto:195411465Zwlj@10.131.2.1">
195411465Zwlj@10.131.2.1</a><br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY<br>Contact: <<a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>><br>Proxy-Authenticate: Digest realm="asterisk", nonce="22a96479"
<br>Content-Length: 0</p>
<p><br>---<br>Scheduling destruction of call <a href="mailto:'195411465Zwlj@10.131.2.1'">'195411465Zwlj@10.131.2.1'</a> in 15000 ms<br>Found user '070001'<br>localhost*CLI><br><-- SIP read from <a href="http://10.131.2.1:5060">
10.131.2.1:5060</a>:<br>ACK <a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://10.131.2.1">10.131.2.1</a>;branch=z9hG4bKaciipncbQ<br>Max-Forwards: 70<br>From: <
<a href="mailto:sip:070001@10.131.0.1">sip:070001@10.131.0.1</a>>;tag=1c1850211233<br>To: <<a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone>;tag=as6a339401<br>Call-ID: <a href="mailto:195411465Zwlj@10.131.2.1">
195411465Zwlj@10.131.2.1</a><br>CSeq: 1 ACK<br>Contact: <<a href="mailto:sip:070001@10.131.2.1">sip:070001@10.131.2.1</a>><br>Supported: em,timer,replaces,path<br>Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
<br>User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006<br>Content-Length: 0</p>
<p><br>--- (12 headers 0 lines)---<br>localhost*CLI><br><-- SIP read from <a href="http://10.131.2.1:5060">10.131.2.1:5060</a>:<br>INVITE <a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone SIP/2.0<br>
Via: SIP/2.0/UDP <a href="http://10.131.2.1">10.131.2.1</a>;branch=z9hG4bKaclMBIpvu<br>Max-Forwards: 70<br>From: <<a href="mailto:sip:070001@10.131.0.1">sip:070001@10.131.0.1</a>>;tag=1c1850211233<br>To: <<a href="mailto:sip:*2@10.131.0.1">
sip:*2@10.131.0.1</a>;user=phone><br>Call-ID: <a href="mailto:195411465Zwlj@10.131.2.1">195411465Zwlj@10.131.2.1</a><br>CSeq: 2 INVITE<br>Proxy-Authorization: Digest username="070001",realm="asterisk",nonce="22a96479" ",uri="
<a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>",algorithm=MD5,response="41cc6e74fc333e770fa28a7db158a495"<br>Contact: <<a href="mailto:sip:070001@10.131.2.1">sip:070001@10.131.2.1</a>><br>Supported: em,100rel,timer,replaces,path
<br>Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE<br>User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006<br>Content-Type: application/sdp<br>Content-Length: 242</p>
<p>v=0<br>o=AudiocodesGW 644554 101011 IN IP4 <a href="http://10.131.2.1">10.131.2.1</a><br>s=Phone-Call<br>c=IN IP4 <a href="http://10.131.2.1">10.131.2.1</a><br>t=0 0<br>m=audio 6070 RTP/AVP 8 0 96<br>a=rtpmap:8 pcma/8000
<br>a=rtpmap:0 pcmu/8000<br>a=rtpmap:96 telephone-event/8000<br>a=fmtp:96 0-15<br>a=ptime:20<br>a=sendrecv</p>
<p>--- (14 headers 12 lines)---<br>Using INVITE request as basis request - <a href="mailto:195411465Zwlj@10.131.2.1">195411465Zwlj@10.131.2.1</a><br>Sending to <a href="http://10.131.2.1">10.131.2.1</a> : 5060 (non-NAT)<br>
Oct 4 13:20:51 NOTICE[4078]: chan_sip.c:5710 check_auth: stale nonce received from '<<a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone>'<br>Reliably Transmitting (no NAT) to <a href="http://10.131.2.1:5060">
10.131.2.1:5060</a>:<br>SIP/2.0 407 Proxy Authentication Required<br>Via: SIP/2.0/UDP <a href="http://10.131.2.1">10.131.2.1</a>;branch=z9hG4bKaclMBIpvu<br>From: <<a href="mailto:sip:070001@10.131.0.1">sip:070001@10.131.0.1
</a>>;tag=1c1850211233<br>To: <<a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone>;tag=as6a339401<br>Call-ID: <a href="mailto:195411465Zwlj@10.131.2.1">195411465Zwlj@10.131.2.1</a><br>CSeq: 2 INVITE
<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY<br>Contact: <<a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>><br>Proxy-Authenticate: Digest realm="asterisk", nonce="0e317db4"
<br>Content-Length: 0</p>
<p><br>---<br>Scheduling destruction of call <a href="mailto:'195411465Zwlj@10.131.2.1'">'195411465Zwlj@10.131.2.1'</a> in 15000 ms<br>Found user '070001'<br>localhost*CLI><br><-- SIP read from <a href="http://10.131.2.1:5060">
10.131.2.1:5060</a>:<br>ACK <a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://10.131.2.1">10.131.2.1</a>;branch=z9hG4bKaclMBIpvu<br>Max-Forwards: 70<br>From: <
<a href="mailto:sip:070001@10.131.0.1">sip:070001@10.131.0.1</a>>;tag=1c1850211233<br>To: <<a href="mailto:sip:*2@10.131.0.1">sip:*2@10.131.0.1</a>;user=phone>;tag=as6a339401<br>Call-ID: <a href="mailto:195411465Zwlj@10.131.2.1">
195411465Zwlj@10.131.2.1</a><br>CSeq: 2 ACK<br>Contact: <<a href="mailto:sip:070001@10.131.2.1">sip:070001@10.131.2.1</a>><br>Supported: em,timer,replaces,path<br>Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
<br>User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.008.006<br>Content-Length: 0</p>
<p><br>--- (12 headers 0 lines)---<br></p></div>
<div><br clear="all"><br>-- <br>Morten Isaksen<br><a href="http://www.misak.dk/blog/">http://www.misak.dk/blog/</a> </div>