anyone with some info on this? thanks again.
On 9/14/05, Erick Perez <[EMAIL PROTECTED]> wrote: > 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, > -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
