[asterisk-bugs] [Asterisk 0016288]: G723 codec has digitzed voice

[Asterisk Bug Tracker]

The following issue has been CLOSED 
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https://issues.asterisk.org/view.php?id=16288 
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Reported By:                globalnetinc
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16288
Category:                   Codecs/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.1.10 
JIRA:                       SWP-593 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2009-11-19 20:04 CST
Last Modified:              2009-12-21 10:11 CST
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Summary:                    G723 codec has digitzed voice
Description: 
I am using an audiocodes mp202b asterisk 1.6.1.10 and a digium tc400b. 
When I call the asterisk box and get prompts all sounds well.  When
asterisk has to bridge the 723 rtp stream to ulaw for the sip provider the
voive becomes very digitzed and poor.

mp202b(723)=> asterisk - works.
mp202b(723)=> asterisk(ulaw)=> sip provider - fails

it produces poor quailty both ways
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 (0115545) lmadsen (administrator) - 2009-12-21 10:11
 https://issues.asterisk.org/view.php?id=16288#c115545 
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Per kpfleming, this appears to be a Digium support issue with the TC400B
board, so I'm going to close this issue and encourage you to contact Digium
support. Sorry for the delay here, as I didn't clue in this was likely a
commercial support issue. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-21 10:11 lmadsen        Note Added: 0115545                          
2009-12-21 10:11 lmadsen        Status                   acknowledged => closed
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