Hello,

On 2/7/11 8:12 PM, Amit Nepal wrote:

I have been trying to figure this out While using kamailio and rtpproxy, the caller is not receiving the bye when callee hangs up but audio is two way and everything seems to be working fine, any one had this issue ?

are you doing record-routing in your config?

The best for providing further hints is to get the SIP trace for such call, from the starting INVITE to the end -- ngrep is recommended to use for sending on this list since it prints out text, following command can be used on your sip server:

ngrep -d any -qt -W byline port 5060

Cheers, Daniel