[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?
> \
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro




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