hi i am using asterisk 1.8.3 i have purchased hosting on go daddy server located in usa.
i have installed asterisk 1.8.3 into the server create 3 extensions 8001,8002,8003. into sip.conf.
here is user details.
same as for 8002,8003. now try to connect all three extension my different 3 offices.make a audio and video call successfully.
but when i have click send video from the soft phone so video is not display give me on asterisk console
rtp codec=null error like
what is the solutions for that?