[Asterisk-Users] Re: Asterisk SIP architecture question

[Kevin P. Fleming]

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.