[OpenSIPS-Users] opensips BYE issue
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Wed Nov 24 12:47:09 CET 2010
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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