[Freeswitch-users] Mod_rtmp

[Amit K Nepal]

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