[Freeswitch-users] Mod_rtmp

Hello all, I am new to freeswitch, trying to get the mod_rtmp working. I am able make call from web client (example) to a sip client, but I am not able to call a web user from another web user , it says user not available. Can you please provide any pointers so I can troubleshoot in the right direction ? I haven't made any changes to the dialplan etc. Thank you Amit