Altus<br><br>It's in the transcoding - <a href="http://www.voip-info.org/wiki-Asterisk+dimensioning">http://www.voip-info.org/wiki-Asterisk+dimensioning</a> 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
<b>20-25</b> decent quality calls before asterisk dies when transcoding and H323 are involved."<br><br>Regards<br>Rob<br><br><div><span class="gmail_quote">On 10/21/05, <b class="gmail_sendername">Altus Snyman</b> <
<a href="mailto:altus@stormcorp.co.za">altus@stormcorp.co.za</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Good day.<br>
I configured asterisk and oh323.Im using it as a sip-h323 convertor<br>A call will come in to the asterisk box via IAX and be send to a quintum<br>h323 gateway.<br>in oh323 you can set the max in,out and simultaneous calls, Ive set them
<br>all to 100.<br>Calls coming in via iax is alaw and then goes out h323 g729.<br>It is a P4 3.3 and 1Gig of ram.Yet at 20+ calls, calls start failing.<br>Is there someone else with a setup like this.Is the problem on the
<br>asterisk side or the quintum<br>Please help<br>Thanks<br>Altus<br>_______________________________________________<br>--Bandwidth and Colocation sponsored by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list
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