[SR-Users] problem with bye using rtpproxy

[Daniel-Constantin Mierla]

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

-- 
Daniel-Constantin Mierla
http://www.asipto.com