Besides logs and any sip traces as Aamir stated it'd be hard to pin point anything. One of the big reason for automatically dropping an established call specially after a specific amount of time i.e 30 seconds..thats because of lack of RTPs flowing through the server and if media-relaying tools are used they will send hangup to both ends, even though both ends have their RTP flowing *directly*.
Regards, Sammy On Tue, Jul 3, 2012 at 12:21 PM, aamir chougule <[email protected]> wrote: > Hi Rodrigo, > > The only which I know is after the call gets connected i.e. 200OK from the > other end there should always be an ACK for that 200OK, if there is no ACK > for the 200OK then the call will get disconnected after sometime. > > Can you please give us the logs for the call by tracing it through ngrep > tool because without any sip traces or opensips log we can't tell what is > going on within the box. > > Regards, > > Aamir Chougule > Cell: 09167989111 > > ------------------------------ > *From:* Rodrigo Ferreira <[email protected]> > *To:* OpenSIPS users mailling list <[email protected]> > *Sent:* Monday, 2 July 2012 10:40 PM > *Subject:* [OpenSIPS-Users] Calls being disconnected > > Hey guys, > > I’m having problems with calls being “disconnected” and I dont know where > I can start to look at, because on my CDR all those calls ends with a > 200OK, but that isnt true, because you are talking all the suddenly the > call stop. > > Any ideas where I should start looking at? > > Engº Rodrigo Ferreira > Supervisor de Telefonia > VIPWay Telecom > Tel.: +55 13 4010-1000 > Cel.: +55 13 8136-5839 > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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