asterisk users - Oct 2004 - Call Quality

Hi

I am having a problem with voice quality getting
bad after about 10-15 minutes on a call. The call
starts out fine but gets very chopy after 10-15 minutes.
It seems like a buffer problem, I am using Asterisk with
a Voicetronix Openlin4 card.

Could someone shed some light on this?

Hi Guys,

I'm new to Asterisk and have managed to setup a server which two sip
clients can log into using xlite. I can successfully call the other
extension and voicemail however I can never hear the other party over
a call or hear a recorded voicemail message.

The little red microphone meters go up and down when you talk and they
definitely are not muted. Could it be some sort of codec issue or
newbie trap? Any tips would be appreciated.

Regards,
Daniel Schell

Something similar:
I have test installation of Asterisk 1.0.RC2 on Pentium1-200MHz, 
Slackware10.
When I'm using ILBC or GSM, same thing happening to me after about 2 min 
conversation.
I'm using g711 on Phone side, and ILBC(GSM) on remote SIP side (internet), 
so Asterisk doing some translation g711-ILBC(GSM) and reverse.

Is there some solution for that behavior?


----- Original Message ----- 
From: "Doug Reid -Stormcorp" <doug@stormcorp.co.za>
To: <asterisk-users@lists.digium.com>
Sent: Wednesday, October 06, 2004 2:53 PM
Subject: [Asterisk-Users] Call Quality

> Hi
>
> I am having a problem with voice quality getting
> bad after about 10-15 minutes on a call. The call
> starts out fine but gets very chopy after 10-15 minutes.
> It seems like a buffer problem, I am using Asterisk with
> a Voicetronix Openlin4 card.
>
> Could someone shed some light on this?
>
>
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