[asterisk-bugs] [Asterisk 0012170]: SIP channel isn't closed when using TLS transport

[noreply at bugs.digium.com]

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:                bbryant
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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:             03-07-2008 15:40 CST
Last Modified:              07-10-2008 08:08 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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---------------------------------------------------------------------- 
 pj - 07-10-08 08:08  
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sorry for delay, but I must wait until issue 0012494 will be resolved, I
haven't pure testing asterisk and issue 0012494 breaks even basic call
functionality for my users. thanks for patience. 

Issue History 
Date Modified   Username       Field                    Change               
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07-10-08 08:08  pj             Note Added: 0089999                          
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