You may want to get in contact with uniza.sk. I believe they are using opensource voip as they have published some excellent tutorials like http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-and-rtpproxy-server even though it is about kamailio and not opensips
On 01/10/2012 07:46 PM, Gabriel Kuri wrote: > Does anyone know of any Universities running OpenSIPS for local call > routing between handsets? > > We're looking at replacing our old Avaya system or upgrade it, and the > forklift upgrade from Avaya is ridiculously expensive (no surprise). > > We'd like to replace our Avaya system with a combination of OpenSIPS > and FreeSWITCH and some Cisco routers for external PSTN access, but > it's going to be a tough sell to our CIO, unless we can show someone > else has done it already. > > Any pointers to other Universities would be great. > > Cheers, > Gabe > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
