[asterisk-bugs] [Asterisk 0014255]: Authentication seems to be broken again for SIP NOTIFY requests

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14255 
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Reported By:                zktech
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14255
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Target Version:             1.6.0.5
Asterisk Version:           1.6.0.1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-01-15 17:15 CST
Last Modified:              2009-01-22 12:26 CST
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Summary:                    Authentication seems to be broken again for SIP
NOTIFY requests
Description: 
This issue is the same as 9896? I am running release version 1.6.0.1 and
the sip notify sends out and I get 401 Unauthorized back This happens with
both Grandstream and Audiocodes devices. I had things working in 1.4 with
the 9896 patch and it was working in the 1.6.0 beta at one point but now it
is back to the pre fix behavior? Any help would be apperciated as I can't
reboot or force config updates without this working.

1695.086809 192.168.150.111 -> 216.109.196.34 SIP Request: NOTIFY
sip:6164581832-10 at 192.168.100.220:31876;transport=udp
1695.114263 216.109.196.34 -> 192.168.150.111 SIP Status: 401
Unauthorized
1697.138730 216.109.196.34 -> 192.168.150.111 SIP Request: REGISTER
sip:65.183.171.213
1697.138889 192.168.150.111 -> 216.109.196.34 SIP Status: 401 Unauthorized
   (0 bindings)
1697.262276 216.109.196.34 -> 192.168.150.111 SIP Request: REGISTER
sip:65.183.171.213
1697.264920 192.168.150.111 -> 216.109.196.34 SIP Status: 200 OK    (1
bindings)
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 (0098467) zktech (reporter) - 2009-01-22 12:26
 http://bugs.digium.com/view.php?id=14255#c98467 
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I am working on this. I will provide examples from asterisk 1.4.17 running
the patch 9896 as well as from the asterisk 1.6.0.1 version.  Do you know
if there is a easy way to save the sip debug info to a log file some where?
Right now I am just cutting and pasting. Will the wireshark traces give you
the sip history you are asking for? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-22 12:26 zktech         Note Added: 0098467                          
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