At 05.23 07/02/2007, you wrote:
Yuan LIU wrote:
After reading through several recent threads, I started to wonder
why the Cisco document (and other VoIP documents) appears to
present this issue as VoIP gateway specific. Don't (plain old)
PBX' face the same issue if they use analogue interfaces? If there
are analogue PBX' at all, how do they solve the problem?
Yes, analog PBXs have the same issues. Don't do anything to solve
the issue. That is way many hotels tell their guests to not let a
call ring for more than 45 seconds or the call will be billed even
if it was not answered.
I agree with LIU. A standard analog PBX tries to solve these billing
problems (for example in Italy you have a "billing pulse" from the
telco that can be intercepted by analog PBX and thus billed). Why
shouldn't Asterisk try to do the same? There's too much confusion
about "call progress" functionality, in Asterisk code and
documentation. Shouldn't be better to say EITHER that it can work in
any country but there's still too much work to do OR that it cannot
work and then take it away from the source code?
I mean if there's a way to make it work (using different systems for
different countries), then I think it's an important feature
(considering also that many companies including Digium sell FXO
module for analog lines). If there is no way, better maybe just get
rid of it and put a red sign on the product specifications of the
analg cards "YOU'LL NOT BE ABLE TO DO BILLING!!!".
Rgds.
Stefano
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