Hi Tomasz,
It looks like the routing set for the call is broken...Could you post
here the SIP capture (from opensips server) of such call (showing in and
out traffic, from the beginning of the call, to the end)
Regards,
Bogdan
Tomasz wrote:
Hello,
Can you helo me with some issue?
I have such scenario:
Dialer registered to asterisk via outbound proxy (TCP) and XLite
connected to the same asterisk via UDP.
When I make a call from dialer to XLite I have no problem until I want
to end a call on XLite side.
When XLite disconnects a call, than on dialer side the call is not
finished.
Wireshark logs show all comunication goes via TCP but BYE is sent from
opensips via UDP.
Is it possible to force opensips to send BYE via TCP too.
I tried:
if (is_method("BYE") && src_ip=="xx.xx.xx.xx")
{
search_append('Request-URI:.*sip:[^>[:cntrl:]]*', ';transport=tcp');
xlog("L_INFO", " $ru \n");
}
but this looks like not working for me.
RURI port and IP is correct but transport is not set to TCP.
Can you help me?
\
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--
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro
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