asterisk users - Jun 2005 - performance of * in several scenarios

Hi,

Is here someone who could provide me any information from practical using of * ?

I need to know more about performance. The main question is:  
"How many extensions should i have configured in and provided with my * box
in several cases":

1. * is used only for SIP signalling, no rtp stream is going through * (always
using reinvite and no nat used in lan/wan)
2. * is used for signalling, rtp stream is going between UAs, but rtp stream is
going through *, when is routed outside (from SIP to TDM world) via SIP trunk
3. * is used for signalling, rtp stream is going between UAs, but rtp stream is
going through *, when is routed outside (from SIP to TDM world) via local
CAPI/ZAP interface
4. * is used for signalling, rtp stream is always going through (never reinvite)

Common details:
- no codec translations  / only one codec used in whole network 
- voicemail system and other  services like wakeup calls, weather information,
...  are running on other (dedicated) * box (hope that its possible)
- no frontend like SER used

Are there any tables or some tools, which could make some calculations for me ?

Thanks, 

B
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Nobody ? :-(

-b
  ----- Original Message ----- 
  From: barney 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Wednesday, June 08, 2005 11:39 AM
  Subject: [Asterisk-Users] performance of * in several scenarios


  Hi,

  Is here someone who could provide me any information from practical using of *
?

  I need to know more about performance. The main question is:  
  "How many extensions should i have configured in and provided with my *
box in several cases":

  1. * is used only for SIP signalling, no rtp stream is going through * (always
using reinvite and no nat used in lan/wan)
  2. * is used for signalling, rtp stream is going between UAs, but rtp stream
is going through *, when is routed outside (from SIP to TDM world) via SIP trunk
  3. * is used for signalling, rtp stream is going between UAs, but rtp stream
is going through *, when is routed outside (from SIP to TDM world) via local
CAPI/ZAP interface
  4. * is used for signalling, rtp stream is always going through (never
reinvite)

  Common details:
  - no codec translations  / only one codec used in whole network 
  - voicemail system and other  services like wakeup calls, weather information,
...  are running on other (dedicated) * box (hope that its possible)
  - no frontend like SER used

  Are there any tables or some tools, which could make some calculations for me
?

  Thanks, 

  B
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