[Asterisk-Users] g729 to asterisk to g729 voip provider

[Erick Perez]

Using sipura sip/g729 to connect to an asterisk server that will
server as a gateway to a VOIP provider, all in g729 will require to
purchase codecs from Digium?

also, in this scenario the transcoding is almost non-existent right?
I have read many documents about the type of codecs, and g729 seems to
be a good trade between almost-toll quality and low bandwith usage
right?


A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can
sustain more than 100 calls or up to a 100?
I just looking at hardware capacity, since the machine will be located
at an ISP with more than needed bandwith.

There is no need for voicemail, web interfaces or anything else, since
the * box will only function as a gateway to a US-based VOIP provider.

The machine in question runs Centos4 Linux (Redhat enterprise 4) and
CDR logging only.

Thanks,