[asterisk-bugs] [Asterisk 0015532]: Getting one way audio after blind transfer from a SIP trunk call.
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jul 21 14:48:47 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15532
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Reported By: olivier1010
Assigned To:
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Project: Asterisk
Issue ID: 15532
Category: Core/RTP
Reproducibility: sometimes
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.4.25.1
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-07-20 05:03 CDT
Last Modified: 2009-07-21 14:48 CDT
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Summary: Getting one way audio after blind transfer from a
SIP trunk call.
Description:
Quite often when calling with a SIP trunk to outside, we loose outgoing
audio.
But we can still hear audio from called party.
We are using G711 alaw for the outgoing SIP trunk.
This appeared with a recent asterisk update to version 1.4.25.1
This is an Elastix 1.5.2 distribution, and the machine is a pentium 4
monocore.
This does not occur with internal calls.
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(0108046) olivier1010 (reporter) - 2009-07-21 14:48
https://issues.asterisk.org/view.php?id=15532#c108046
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After some deep investigations, we have found that we have adding a new SIP
trunk to the system a few days ago who needed a different defaultexpirey
value of 1800. (for a provider that is refusing lower than this value).
Because of this new defaultexpirey, other SIP trunks were loosing
registration.
According to the documentation, it is not possible to specify a different
defaultexpirey for each user / peer.
We solved the problem using static SIP trunks.
A defaultexpirey by user / peer should be implemented to avoid those
problems.
Issue History
Date Modified Username Field Change
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2009-07-21 14:48 olivier1010 Note Added: 0108046
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