Kevin, Is the CDR accounting done based on SIP signaling? If a UA is talking (RTP) to a third party PSTN gateway, isn't it at risk if say the UA loses power. How will asterisk know the call has ended if it is not involved in the media path. The idea is this.. I want to use canreinvite =yes to force users to talk end-to-end to preserve bandwidth, but I can see the potential for hung calls if asterisk never get the BYE from a UA in the event the ATA gets unplugged or somehow loses power.
regards David On 11/23/05, Kevin P. Fleming <[EMAIL PROTECTED]> 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 :-( > > SIP re-INVITEs do _not_ affect the CDRs in any way, period. They only > affect the media streams. > _______________________________________________ > --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
