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 oh
[mailto:[EMAIL PROTECTED] Behalf Of Altus Snyman
Sent: Friday, October 21, 2005 1:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] how many oh323
Good day.
I configured asterisk and oh323.Im using it as a sip-h323 convertor
A call will come in to the asterisk
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 on
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 inv
Good day.
I configured asterisk and oh323.Im using it as a sip-h323 convertor
A call will come in to the asterisk box via IAX and be send to a quintum
h323 gateway.
in oh323 you can set the max in,out and simultaneous calls, Ive set them
all to 100.
Calls coming in via iax is alaw and then goe
