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

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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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:                     new
Asterisk Version:           1.6.1.10 
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:             2009-11-19 20:04 CST
Last Modified:              2009-11-19 20:04 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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Issue History 
Date Modified    Username       Field                    Change               
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2009-11-19 20:04 globalnetinc   New Issue                                    
2009-11-19 20:04 globalnetinc   Asterisk Version          => 1.6.1.10        
2009-11-19 20:04 globalnetinc   Regression                => No              
2009-11-19 20:04 globalnetinc   SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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