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.
[general]
videosupport=yes
[8001]
type=friend
username=8001
secret=hidden
host=dynamic
context=from-camera
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=speex
allow=gsm
allow=h261
allow=h263
allow=h263p
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?