[asterisk-users] Disconnection supervision: what about PBX

[Stefano Corsi]

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