[asterisk-bugs] [Asterisk 0014776]: [Help] no voice after call get connected in g729 codec

Asterisk Bug Tracker noreply at bugs.digium.com
Sun Mar 29 00:11:52 CDT 2009


The following issue has been RESOLVED. 
====================================================================== 
http://bugs.digium.com/view.php?id=14776 
====================================================================== 
Reported By:                balasam
Assigned To:                russell
====================================================================== 
Project:                    Asterisk
Issue ID:                   14776
Category:                   Codecs/General
Reproducibility:            random
Severity:                   major
Priority:                   normal
Status:                     resolved
Asterisk Version:           1.4.24 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 suspended
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-03-28 13:34 CDT
Last Modified:              2009-03-29 00:11 CDT
====================================================================== 
Summary:                    [Help] no voice after call get connected in g729
codec
Description: 
I setup a asterisk environment with the sip trunk and the codec used is
g729. For some of the  calls, after call connect there is no voice and it
is  blank. This is happening for 30% precentage of calls. I written a
dialer Application in AMI, it will make the outbound call with one
particular number and the call will be connected to the available SIP
extension.
 
Asterisk - 1.4.24.


I purchased 151 licenses of g729.
Specification : 64bit
codec_g729a-1.4_3.0.3-nocona.tar.gz
Ouptut of show g729 CLI command.
> show g729
2/4 encoders/decoders of 151 licensed channels are currently in use.

Output of show translation CLI command,
>
> show translation
Translation times between formats (in milliseconds) for one second of
data
Source format (Row) Destination format (Columns)
g723 gsm ulaw alaw g726aa12 adpcm slin lpc10 g729 speex ilibc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 2 2 2 2 1 4 9 - - 2 -
ulaw - 2 - 1 2 2 1 4 9 - - 2 -
alaw - 2 1 - 2 2 1 4 9 - - 2 -
g7266aa12 - 2 2 2 - 2 1 4 9 - - 1 -
adpcm - 2 2 2 2 - 1 4 9 - - 2 -
slin - 1 1 1 1 1 - 3 8 - - 1 -
lpc10 - 3 3 3 3 3 2 - 10 - - 3 -
g729 - 2 2 2 2 2 1 4 - - - 2 -
speex - - - - - - - - - - - - -
ilibc - - - - - - - - - - - - -
g726 - 2 2 2 1 2 1 4 9 - - - -
g722 - - - - - - - - - - - - -



and the phone which i am using is Cisco 7940 and it support the g729
codec. The problem is 30% of calls are no voice calls. I am using the
FreePBX for asterisk administration. Please Kindly direct me in right
direction.

Thanks in  Advance,
balasam
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-29 00:11 russell        Resolution               open => suspended   
2009-03-29 00:11 russell        Assigned To               => russell         
======================================================================




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