[Asterisk-Users] Asterisk SIP architecture question

[Kevin P. Fleming]

Matt Riddell wrote:

> The fact remains, if you need *very* accurate cdr's then you either don't do
> canreinvite=yes for the peer or you code something so that Asterisk notices
> that the rtp has stopped.  The fact remains that without these, the most
> accurate CDR is going to come from the provider.

OK, I'll agree with that, except for one thing: if the provider notices 
that the RTP has stopped and wants to kill the call, it will send BYE to 
Asterisk and Asterisk will close the channels and update the CDR. The 
only time this will be an issue is if _both_ ends disappear and never 
send any signaling to Asterisk.

However, in the general case of not being concerned so much about the 
peer going away and losing CDR information for _one_ call, using 
reinvites does _not_ impact the quality of the softswitch's (Asterisk) CDRs.