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? regards, David
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)
Kevin P. Fleming
2005-Nov-23 09:08 UTC
[Asterisk-Users] Asterisk SIP architecture question
Matt Riddell wrote:> 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 CDRsSorry, this advice is bogus :-( SIP re-INVITEs do _not_ affect the CDRs in any way, period. They only affect the media streams.
David Thomas
2005-Nov-23 09:17 UTC
[Asterisk-Users] Re: Asterisk SIP architecture question
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 <matt.riddell@sineapps.com> 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 >
Kevin P. Fleming wrote:> Matt Riddell wrote: > >> 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 > > > Sorry, this advice is bogus :-(So how does Asterisk know that the media stream has been disconnected between the two remote hosts? -- 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)