[asterisk-bugs] [Asterisk 0015466]: rtpkeepalive option doesn't always work
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Sep 18 09:25:50 CDT 2009
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=15466
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Reported By: viraptor
Assigned To:
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Project: Asterisk
Issue ID: 15466
Category: Channels/chan_sip/General
Reproducibility: sometimes
Severity: minor
Priority: normal
Status: closed
Asterisk Version: 1.6.1.1
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: no change required
Fixed in Version:
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Date Submitted: 2009-07-08 10:05 CDT
Last Modified: 2009-09-18 09:25 CDT
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Summary: rtpkeepalive option doesn't always work
Description:
I've tried enabling rtpkeepalive option (2 seconds) in a server without nat
or any specific dialplan (just a Dial(SIP/...) ). It works fine on one
server, but doesn't on the other. I cannot find any reason for it though.
Configs are almost the same. The only real difference between them is that
one is multihomed (the one where rtpkeepalive fails), but I'm not sure
that's related at all.
All audio passes through asterisk in both setups.
I expect the rtpkeepalive to be sent while the call is on hold.
How can I debug this issue?
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(0110981) lmadsen (administrator) - 2009-09-18 09:25
https://issues.asterisk.org/view.php?id=15466#c110981
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I'm going to close this issue for now because this feels more like a
support issue (or at least the very early stages of bug triage) where it
would make more sense to get help and information from the asterisk-users
mailing list. If you gather enough information to determine where (or if)
the bug exists, then please feel free to reopen this issue. Thanks!
Issue History
Date Modified Username Field Change
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2009-09-18 09:25 lmadsen Note Added: 0110981
2009-09-18 09:25 lmadsen Status new => closed
2009-09-18 09:25 lmadsen Resolution open => no change
required
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