Hi Klaus,
What U 'd told , I did it.
but i didn't see the Bye Request to server , By using both ngrep and
ethereal..
On 7/21/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
How should I know?
Do what I suggested. Use ngrep (or any other packet sniffer like tcpdump
or ethereal/whireshark) to watch where the BYE gets lost. Then I can
help.
btw: please send emails to the list
regards
Klaus
raviprakash sunkara wrote:
Hello Klaus,
thank U..... for replying to me..
CAN help On Bye Problem
is openser is problem or network problem..
can do that...
Bye
On 7/20/06, * Klaus Darilion* <klaus.mailinglists(a)pernau.at
<mailto: klaus.mailinglists(a)pernau.at>> wrote:
raviprakash sunkara wrote:
>
> Hi Users
>
> I'm Using the openser with Nat Bu using the RTP..
>
> After invite method rtp is also established between the
caller and
callee
...
Audio is clear..
But when I send Bye request to Callee , it not hung upping to
callee it
still estallishing the call...
I think that ....
1) openSER server is not getting the
request from
> callee or caller
> 2) OPenser is respones the BYE ..
> 3) Router(firewall) gateways is
blocking the
> Bye request , to pass the request to
openser,
> 4) Route behind the UA's
isbloacking....
You have to check which one of this is the real case.
Use ngrep at the SIP proxy to verify if the BYE is received and
forward.
Take a look at the IP addresses and ports in
the Route headers and
request URI and Contact headers. Make sure the BYE is sent to the
proper
IPaddress:socket.
regards
klaus
--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
Kondapur, Hi-tech city,
Hyderabad.
www.hyperion-tech.com <http://www.hyperion-tech.com>
+91-9985077535
--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
Kondapur, Hi-tech city,
Hyderabad.