[asterisk-bugs] [Asterisk 0012170]: SIP channel isn't closed when using TLS transport
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jul 31 14:01:06 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12170
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Reported By: pj
Assigned To:
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Project: Asterisk
Issue ID: 12170
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: ready for testing
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 104031
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 2008-03-07 15:40 CST
Last Modified: 2008-07-31 14:01 CDT
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Summary: SIP channel isn't closed when using TLS transport
Description:
when H323 endpoint calls SIP, call is established and then h323 hangs up,
asterisk doesn't send sip BYE to sip endpoint and thus channel remains open
until RTP times out.
when I tried to setup call in oposite direction, ie. sip endpoint calls
h323, call is established, then h323 hangs up, sip BYE is send and channel
is correctly closed.
I'm not observing this issue, when using udp as sip signaling transport.
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Relationships ID Summary
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has duplicate 0012700 Zaptel channel detects hangup, but does...
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(0090935) pj (reporter) - 2008-07-31 14:01
http://bugs.digium.com/view.php?id=12170#c90935
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I don't know, if lock was caused by your patch.
But tls seems to be somehow buggy, I have more issues with tls, eg.
http://bugs.digium.com/view.php?id=13117
so, If you think, that your patch solves issue from this bugreport (not
sending BYE to end call), you can try to commit and close this bugreport.
When tls will better usable later, I will try again and if 'BYE issue'
will appear again, I will reopen this bugreport.
Issue History
Date Modified Username Field Change
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2008-07-31 14:01 pj Note Added: 0090935
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