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 PM *To:* TAUG *Subject:* [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
