Hi
I have an asterisk server on public ip and two sip clients registered on it…when i try call from client A to B after 30 seconds call hangs up automatically similary calling from B to A clients hangup after 30 seconds.
Sip.conf
[786]
username =786
host=dynamic
type = friend
NAT = route
secret=****
context=default
callerid=786
Any help would be highly appreciated…Thanks alot
do a packet capture on port 5060 on your asterisk box. Make sure that 200OK is making it from one phone to the other, and then make sure that the ACK makes it back as a response. I believe the timeout for that no ACK back is 30 seconds.
Also, your phones may be re-inviting. Does the call actually drop, or does just the audio drop?
thanks alot
problem is solved by changing bind port to 9060 .