I was thinking in Kamailio, but this sip proxy handles only the SIP signalling traffic, no media processing.
On 3 November 2011 17:07, Nick Khamis <sym...@gmail.com> wrote: > Shouldn't you be using a Proxy? > > Nick. > > On Thu, Nov 3, 2011 at 1:04 PM, Sunny <no7f...@gmail.com> wrote: > > Hi list, > > Could anyone tell me what is the "recommended" hardware to a system for > > following configuration: > > SBC --> Asterisk (SS) --> Carrier GW > > Asterisk should work as a Class 4 SoftSwitch, with following > functionalists: > > -> Do the IP Authentication > > -> All communications on RTP/G729 (no transcoding required) > > -> Load of 1200 concurrent call sessions > > -> No call routing required > > Thanks in advance, > > Sunny > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
