Kevin P. Fleming wrote: > David Thomas wrote: > >> 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. > > > That is the case in every SIP network, Asterisk is not unique in that > regard. > > I would suggest that you could make a modification to chan_sip so that > if the peer goes 'unreachable' (as determined by using qualify=yes) than > any existing calls involved with that peer are immediately hung up; that > would take care of this problem.
That would be a good addition. Optional of course. /O _______________________________________________ --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