As for with Trixbox, I know that there are problems reported with G729 from 
digium and there are suggestions of using the free Intel codecs. They worked 
for my box without any problem.


Date: Wed, 8 Nov 2006 16:33:39 -0500From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: 
[EMAIL PROTECTED]: Re: [on-asterisk] Asterisk call and disconnecting after 
50-60 sec
HI,
 
Thank you.
 
I have 20 Licenses from digium.
 
CPU usage is < 10%
 
Thanks
Lloyd 
On 11/8/06, Bjorn Asmul <[EMAIL PROTECTED]> wrote: 


Hi Lloyd,
 
First of all you need the proper g729 codec (& license) installed in order to 
transcode to/from g729.
Check to see if the call is actually transcoded.
You should have g729 on one leg, and g711/ulaw on the other.
If you never have different codecs in use for both legs, chances are you don't 
have the codec installed or enabled.
You can check this by running "show translation" within Asterisk CLI.
 
g729 is a lossy codec, and is not as robust as g711 when used over poor 
connections (high latency, jitter and other bad things). 
 
g729 is also heavier on your CPU, when doing transcoding, so you might want to 
check your CPU usage when the call drops.
 
All this is also true for any of your end-points, whether its your provider or 
end-user.
 
 
Thanks,
Bjorn Asmul
www.atlasvoice.com 


From: Aloysius Thevarajah Lloyd [mailto:[EMAIL PROTECTED] Sent: Wednesday, 
November 08, 2006 3:00 PMTo: TAUGSubject: [on-asterisk] Asterisk call and 
disconnecting after 50-60 sec 

Hi,
 
 
Could some one help me to troubleshooting call disconnection problem
 
Customer ( Make Dial DID SIP/g711 ) <=> Asterisk <=> Dial Destination ( g729) 
 
Not every time some times Asterisk Lost the Destination ( g729) Connection. I 
tried destination different providers.
 
Also I am doing a transcoding insdie the box. 
 
But I never lose the connection between  Customer ( Make Dial DID SIP/g711 ) 
<=> Asterisk.
 
Any suggestion.
 
Thank you.
LLoyd
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