[asterisk-bugs] [Asterisk 0014255]: Authentication seems to be broken again for SIP NOTIFY requests
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Nov 25 15:38:29 CST 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/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: new
Target Version: 1.6.0.5
Asterisk Version: 1.6.0.1
JIRA:
Regression: No
Reviewboard Link:
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: 2009-11-25 15:38 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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(0114296) OwenL (reporter) - 2009-11-25 15:38
https://issues.asterisk.org/view.php?id=14255#c114296
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More information - the NOTIFY goes out and receives a 401 back asking for
authorization. The packet is received in handle_response in chan_sip.c but
is dropped when __sip_ack sets ack_res false.
It appears the packet is never looked at thus the authorization code is
never reached.
This was the case the 1.6.1 and 1.6.2 versions.
SIP debug, Call history, and debug logs are attached in a single file
I'm new to the code so am not sure what the proper solution is. It would
be very useful for the Linksys phones we are using.
Thanks!
Issue History
Date Modified Username Field Change
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2009-11-25 15:38 OwenL Note Added: 0114296
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