I know this may sound like a stupid question but I will put on my flame
retardant suit and ask anyway.
Is there any way to use/allow SIP reinvite and still track the length of the
call?
I realize that the whole idea of reinvite is that it takes the proxy out of
the media path which, from what I understand also kills the proxy's ability
to track the start/end time of the call for billing purposes.
Are there any really smart guys out there with propeller hats that have come
up with a way to get the best of both worlds?
Do we lose anything else using reinvite with Asterisk?
Thanks in advance for any help..
--Mojo
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