[asterisk-bugs] [Asterisk 0018798]: Call drop with chan_sip
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Mar 7 09:06:59 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18798
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Reported By: agustina
Assigned To:
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Project: Asterisk
Issue ID: 18798
Category: Channels/chan_sip/General
Reproducibility: unable to reproduce
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: Older 1.6.2 - please test a newer version
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2011-02-11 13:27 CST
Last Modified: 2011-03-07 09:06 CST
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Summary: Call drop with chan_sip
Description:
I have the following arquitecture:
Asterisk with 2 E1, both are PCI express, so there wouldn`t be IRQ
issues.
I have my Asterisk 1.6.2.13 conected to a switch and my softphones and
hardophones connected to another switch.
What happens is that calls drop suddenly ramdomly. We have some sound
quality issues but don`t see errors other than some:
channel.c: Dropping incompatible voice frame on SIP/XXXX-0000326f of
format alaw since our native format has changed to 0x4 (ulaw),
but this messages happen at all times and not when the call drop happens.
We are suspecting some problem with SIP protocol.
We are ataching full and the part where the call hangups.
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(0132691) agustina (reporter) - 2011-03-07 09:06
https://issues.asterisk.org/view.php?id=18798#c132691
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Folks!
What we find was this parameter was set (in chan_dahdi.conf):
callprogress=yes
(; This feature can also easily detect false hangups. The symptoms of this
is being disconnected in the middle of a call for no reason.)
this parameter we commented and we also commented:
busydetect=yes
busycount=10
and call drops stopped.
Issue History
Date Modified Username Field Change
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2011-03-07 09:06 agustina Note Added: 0132691
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