[asterisk-bugs] [Asterisk 0016288]: G723 codec has digitzed voice
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Dec 21 09:58:32 CST 2009
The following issue has been UPDATED.
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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: acknowledged
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: 2009-12-21 09:58 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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2009-12-21 09:58 lmadsen version 1.6.1.10 =>
2009-12-21 09:58 lmadsen Description Updated
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