<div>HI Lists,</div> <div> </div> <div>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.</div> <div> </div> <div>Currently i am using Asterisk as Softswitch to route the calls to voip carriers, accounting and athurization.</div> <div> </div> <div>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.</div> <div> </div> <div>Is FS support kind of realtime database so we can let SIP ATAs register direct from database?</div> <div>Does it have some API so we can run Perl to calculate max credit time for priticular destinations?</div> <div> </div> <div>I will be appriciate if any one can guide me in right way so i can start from there?</div> <div> </div> <div>Thank You</div> <div>Abdul Lateef</div><p> 
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