[asterisk-bugs] [Asterisk 0010443]: One way totaly distorted audio, the other way is OK.
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Aug 13 18:18:16 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=10443
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Reported By: phokz
Assigned To: dbowerman
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Project: Asterisk
Issue ID: 10443
Category: Addons/chan_mobile
Reproducibility: random
Severity: major
Priority: normal
Status: assigned
Asterisk Version: 1.4.10
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 424
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 08-13-2007 17:02 CDT
Last Modified: 08-13-2007 18:18 CDT
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Summary: One way totaly distorted audio, the other way is OK.
Description:
My test setup is
Nokia6234---BT Dongle MSI BToes---openSUSE 10.2 w/Asterisk
1.4.10---Twinkle sip phone on same machine.
When I place call from SIP to mobile, it dials and with 70% probability
everything goes just fine.
However, some calls are bad. The audio from phone do not get through to
pbx. SIP phone plays back random memory content - loud noise. In syslog
appears message
Aug 13 23:57:56 gromit kernel: hci_scodata_packet: hci0 SCO packet for
unknown connection handle 73
just before call is bridged. It looks as if receiving thread died or
somehow blocked. There must be a kind of race condition.
But sometimes the same messages appear in log and call is good.
First call after plugging in BT dongle is almost certainly good. Once
there is a bad call, probability of next call being bad too is high. After
unplugging/replugging BT dongle the next call is almost certainly good.
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dbowerman - 08-13-07 18:18
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phokz:
could you please attach the console output when this happens.
be sure to 'core set debug 1' and make sure debug output goes to the
console via /etc/asterisk/logger.conf
Issue History
Date Modified Username Field Change
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08-13-07 18:18 dbowerman Note Added: 0068803
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