[asterisk-bugs] [Asterisk 0010824]: Poor audio quality and sometimes disconnection after hangup

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10824 
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Reported By:                swatchy
Assigned To:                dbowerman
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Project:                    Asterisk
Issue ID:                   10824
Category:                   Addons/chan_mobile
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!): 83834 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             09-26-2007 03:18 CDT
Last Modified:              12-27-2007 06:08 CST
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Summary:                    Poor audio quality and sometimes disconnection after
hangup
Description: 
Some information about my system:
OS Ubuntu 7.04
Bluez: 3.9-0ubuntu1
Bluetooth: Internal Bluetooth from Acer C110 Notebook
Asterisk: SVN-trunk-r83834
chan_mobile: Revision 451
Cellphone: Nokia 6230i

These things are working:
1. Notebook and Nokia are paired
2. Outgoing call from iax phone -> asterisk -> pstn works
3. Incoming call not tested
4. Incoming/outgoing SMS are working

The problem:
At the time the pstn phone picks up the call from the cellphone, you can
hear the person sitting behind the iax softphone very quiet and with a
really poor audio quality. It sounds like a robot is speaking to you. 

My tries to solve the problem:
In mobile.conf:
forcemaster set to no and yes
alignmentdetection set to no and yes
Set the iax softphone to gsm and ulaw

Another problem:
Sometimes after hangup the call the connection between asterisk and the
cellphone breaks up. I don't know why? See this for more information:






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---------------------------------------------------------------------- 
 latysheff - 12-27-07 06:08  
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I connect to my test asterisk by sip from another asterisk box.
If sip mode set to inband, voice disappears after digit is pressed.
If sip mode set to rfc, tone 'hangs' (regenerating permanently).

But if I comment out (again :)) one string in chan_mobile.c, everithing is
ok:

                    ast_dsp_set_features(pvt->dsp,
DSP_FEATURE_DTMF_DETECT);
                    //ast_dsp_digitmode(pvt->dsp, DSP_DIGITMODE_DTMF |
DSP_DIGITMODE_RELAXDTMF);
                    pvt->adapter = adapter;

A bit annoying thing is long tones (http://bugs.digium.com/view.php?id=5#1 sec)
generated by Asterisk into
Mobile channel, but perhaps it is a subject for configuring. 

Issue History 
Date Modified   Username       Field                    Change               
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12-27-07 06:08  latysheff      Note Added: 0076012                          
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