[Asterisk-Users] Re: Asterisk SIP architecture question

[David Thomas]

Kevin,

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.

regards
David

On 11/23/05, Kevin P. Fleming <kpfleming at digium.com> wrote:
> Matt Riddell wrote:
>
> > No, not accurately.  Asterisk may not receive any information in this
> case.
> > The best bet is that if you are doing reinvite to make an agreement with
> your
> > VoIP provider to get a copy of their CDRs
>
> Sorry, this advice is bogus :-(
>
> SIP re-INVITEs do _not_ affect the CDRs in any way, period. They only
> affect the media streams.
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