HI Lists, This is my first post in this list i hope i will get the guide and information to help us moving from asterisk to freeswitch. Currently i am using Asterisk as Softswitch to route the calls to voip carriers, accounting and athurization. But some time i am getting problem in CDRs the calls hanged in asterisk without any RTP and audio. For this reason i want to switch to freeswitch if it is right choise for softswitch purpose. Is FS support kind of realtime database so we can let SIP ATAs register direct from database? Does it have some API so we can run Perl to calculate max credit time for priticular destinations? I will be appriciate if any one can guide me in right way so i can start from there? Thank You Abdul Lateef --------------------------------- Looking for last minute shopping deals? Find them fast with Yahoo! Search.