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
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