[Asterisk-Users] How to increase the performance?

[Rich Adamson]

> >>In the past I had problems with the audio over sip. Then I tried the 
> >>"-p" Option and increased the memory. Now it is better but not perfect.
> >>
> >>Are there any more possibilities to increase it more? By now I'm using a 
> >>P-II/333.
> >>
> >>Could a completely hand optimized kernel (I use 2.6.) help a bit?
> > 
> > There's no way to answer your question with any degree of reasonable
> > truth as you haven't mentioned they type of phones, type of pstn interface,
> > which codecs, etc, etc. 
> 
> Okay. My server has got:
> 
> - One Phonejack Lite
> - One X100P Clone
> - 256mb Memory
> - P II/333
> - Linux 2.6.5
> - Debian Woody
> - Asterisk 1.0.1
> - Codecs: GSM, ulaw, alaw
> - ADSL 1000kBit/s Downstream, 128kBit/s Upstream
> 
> Calls from or to the pstn are completely okay. Calls over SIP aren't. 
> Calls over IAX couldn't be tested at the moment.
> 
> 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?

Current version of * or what?

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?