[asterisk-bugs] [Asterisk 0014550]: Rtp socket are not closed after Hangup
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Feb 25 04:46:02 CST 2009
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=14550
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Reported By: triccyx
Assigned To:
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Project: Asterisk
Issue ID: 14550
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: SVN
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-02-25 04:36 CST
Last Modified: 2009-02-25 04:46 CST
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Summary: Rtp socket are not closed after Hangup
Description:
After a sip call the rtp ports are never closed.
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(0100697) triccyx (reporter) - 2009-02-25 04:46
http://bugs.digium.com/view.php?id=14550#c100697
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I only use autocreated peers.
In the full log there are some log of mine you can ignore them.
portlog cames from "netstat -lnp | grep asterisk > /usr/src/portLog"
The test has been done with 18 simpleopan phones that do loop calls
Issue History
Date Modified Username Field Change
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2009-02-25 04:46 triccyx Note Added: 0100697
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