AltusIt's in the transcoding - http://www.voip-info.org/wiki-Asterisk+dimensioning has some notes on oh323 v.s. chan_h323 (chan_h323 is just pass through) - someone says there that you won't be able to run more than
20-25 decent quality calls before asterisk dies when transcoding and H323 are
I've been thinking of using yate
http://yate.null.ro/pmwiki/index.php/Main/H323ToSIPSignallingProxy to do
this. Any thoughts or experiences?
Darren Wiebe
[EMAIL PROTECTED]
Rob Lith wrote:
Altus
It's in the transcoding -
http://www.voip-info.org/wiki-Asterisk+dimensioning has some notes
Altus,
Just looking over the voip-info wiki
http://www.voip-info.org/wiki/Asterisk+dimensioning it seems you have hit
the limit of h323.
about 1/3 way down won't be able to run more than 20-25 decent quality
calls
Shawn
-Original Message-
From: [EMAIL PROTECTED]
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oops, typo!
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
-Original Message-
From: Shawn Porter [mailto:[EMAIL PROTECTED]
Sent: Friday, October 21, 2005 10:08 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] how many oh323
Altus
