[asterisk-bugs] [Asterisk 0019182]: [patch] [regression] Asterisk drops sip messages and/or response codes if SIP/TLS is used
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon May 23 09:29:41 CDT 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=19182
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Reported By: st
Assigned To: mnicholson
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Project: Asterisk
Issue ID: 19182
Category: Channels/chan_sip/TCP-TLS
Reproducibility: always
Severity: block
Priority: normal
Status: closed
Target Version: 1.8.5
Asterisk Version: 1.8.3.3
JIRA: SWP-3404
Regression: Yes
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: fixed
Fixed in Version: 1.8.5
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Date Submitted: 2011-04-26 10:11 CDT
Last Modified: 2011-05-23 09:29 CDT
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Summary: [patch] [regression] Asterisk drops sip messages
and/or response codes if SIP/TLS is used
Description:
When a Snom 360 (Firmware 7.3.7) tries to register, there is a new tcp
connection. Sometimes Asterisk CLI shows Register and 401 Response, but the
phone does not get a response. Sometimes CLI shows nothing, but the phone
logs many attempts to register. Its strange and not easy to reproduce, but
here sipp is the solution.
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Relationships ID Summary
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parent of 0019192 [patch] [regression] segfault in _sip_t...
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(0135254) svnbot (reporter) - 2011-05-23 09:29
https://issues.asterisk.org/view.php?id=19182#c135254
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Repository: asterisk
Revision: 320221
U tags/1.8.4.1/ChangeLog
U tags/1.8.4.1/channels/chan_sip.c
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r320221 | lmadsen | 2011-05-23 09:29:39 -0500 (Mon, 23 May 2011) | 1 line
Merge changes for issue https://issues.asterisk.org/view.php?id=19182 and update
the ChangeLog.
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http://svn.digium.com/view/asterisk?view=rev&revision=320221
Issue History
Date Modified Username Field Change
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2011-05-23 09:29 svnbot Checkin
2011-05-23 09:29 svnbot Note Added: 0135254
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