Hello, just to let everyone know that I updated my tutorial about using Kamailio SIP Server and FreeSwitch for building telephony systems. This time I used latest stable branches from the two projects. The tutorial is available online at: * http://asipto.com/u/k33fs It follows the same concept of using Kamailio for handling authentication, registration and routing logic, with FreeSwitch providing the media services (voicemail, conferencing, ivr, ...) and SBC functionality. One thing promised with the previous version was ability to generate user directory automatically from Kamailio's subscriber table. It is done now for voicemail using a Lua script that connects to Kamailio's database and builds the appropriate XML directory file. Hope is going to be useful for many out there! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu