[asterisk-bugs] [Asterisk 0010443]: One way totaly distorted audio, the other way is OK.

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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:              08-14-2007 01:45 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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---------------------------------------------------------------------- 
 phokz - 08-14-07 01:45  
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dbowerman: log is attached. I made 4 calls, first three calls were good,
last call was bad.

I'd like to use chan_mobile in production environment with sip, iax2 and
isdn channels. Thats why I'd prefer to make it working with 1.4.10.

I also tested svn version of asterisk a while ago, but as development
version might be broken, once I tried it it just crashed, second time I got
no sound at all. I should give it a try again. 

Issue History 
Date Modified   Username       Field                    Change               
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08-14-07 01:45  phokz          Note Added: 0068808                          
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