[asterisk-bugs] [Asterisk 0012170]: SIP channel isn't closed when using TLS transport
Asterisk Bug Tracker
noreply at bugs.digium.com
Sat Aug 2 04:18:16 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: 2008-08-02 04:18 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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(0091012) pj (reporter) - 2008-08-02 04:18
http://bugs.digium.com/view.php?id=12170#c91012
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I'm afraid, that 'transport' doesn't work as expected...
I have only one transport permited (transport=udp) in sip.conf peer
definition, but I'm still able to successfully register and even make calls
using tls. Only some warnings are printed on cli, during peer registration,
but after some attempts, peer is registered. Same behaviour, when I have in
asterisk transport=tls only, peer can be successfully registered via UDP.
cli.txt attached.
Issue History
Date Modified Username Field Change
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2008-08-02 04:18 pj Note Added: 0091012
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