Thanks for the information Matt! Does asterisk store any SIP dialog cdr info in mysql like Call-ID & Cseq? With This info I could at least detect runaway calls and fake a BYE to the pstn gateway with an external app.
regards, David On 11/23/05, Matt Riddell <[EMAIL PROTECTED]> wrote: > David Thomas wrote: > > When asterisk is setup to allow SIP users to send media end-to-end > > (canreinvite=yes), can cdr info still be reliable, considering one of > > the end-user devices could go down leaving the call open. This is > > assuming you are using a third party pstn and not asterisk for pstn. > > > > Does asterisk have any mechanism for detecting and disconnecting hung > > calls in this type of scenario? > > No, not accurately. Asterisk may not receive any information in this case. > The best bet is that if you are doing reinvite to make an agreement with > your > VoIP provider to get a copy of their CDRs > > -- > Cheers, > > Matt Riddell > _______________________________________________ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://freevoip.gedameurope.com (Free Asterisk Voip Community) > http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users