[Asterisk-Users] How to increase the performance?

[Michael Vogel]

Rich Adamson schrieb:

>>Do you need any more facts?
> 
> Sure, getting closer...
> 
> Help us understand what "calls over sip" means. From what device to what
> device when the call is bad (need to understand the path that you're
> talking about including any transcoding going on (if any), what type
> of sip phone, is the sip connection local or through the dsl, and the
> other end of this 'bad call' where is it?

Okay. The call goes to or from a phone connected to the Phonejack Lite. 
The Asterisk Server (Version 1.0.1) converts from signed linear 
(Phonejack) to ULAW/ALAW/GSM. Then it is transferred over ADSL to 
sipgate.de (ping time 100ms) (most sip-calls are incoming calls from the 
pstn to the sipgate.de-gateway)

> Current version of * or what?

No. It's the 1.0.1 since there are no newer versions available for 
debian woody.

> Where ever this sip connection goes that is you're referring to, are
> there any CLI errors or have you tried to use a packet sniffer to
> see what's going on?

I haven't used any packet sniffer by now. The sound problems are 
occuring in 10-30% of the time. So I guess this can only be a 
performance or traffic problem. But since the sound problems are 
occuring in both directions I guess its a performance problem.

Bye!

Michael