[Asterisk-Users] Re: Asterisk SIP architecture question

[Olle E. Johansson]

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