[asterisk-bugs] [Asterisk 0016288]: G723 codec has digitzed voice
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Dec 21 10:11:21 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: 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: no change required
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
2009-12-21 10:11 lmadsen Resolution open => no change
required
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