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.
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