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
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 involved."
Regards
Rob
On 10/21/05, *Altus Snyman* < [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
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 goes out h323 g729.
It is a P4 3.3 and 1Gig of ram.Yet at 20+ calls, calls start failing.
Is there someone else with a setup like this.Is the problem on the
asterisk side or the quintum
Please help
Thanks
Altus
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