A dump question?

In a calling card circumstance, user calls a local access # routes to Asterisk box, and asterisk routes call to another destination #. My question is the local access # be occupied or not even after the call has been routed to destination # ?

Explain what you’re trying to do a bit better. Will the calls take place over analog lines? Will they be over an internet connection?

In the states, if the calling card user you describe was using an analog line on the PSTN to call the Asterisk server, that line would remain occupied for the duration of the call, no matter where the Asterisk server subsequently connected the call to. I would imagine the phone lines in most other countries would have the same restriction.

SIP users can connect to one another after the inital connection has been established. However, in a calling card situation, you’d likely want to disable this capability to control billing.

Yes, users will use analog phone to dial an access # that has been configured with Asterisk box to take incoming calls, then transfer the calls to destination #. So what you are saying is the access # line remains busy during the whole calling process? So how is calling card company doing the business? do they have to use a lot access # if they have a lot customers to make calls at the same time?

My guess is you’re correct, they likely have a number of lines.

That said, there are arrangements that one can set up with the phone company wherein you are merely the entity that distributes calling cards and all of the switching is done at the CO. I don’t know what role an Asterisk server would play in this sort of arrangement.

If you have a pstn number you would have many lines and the teleco does roll over, when they call in the line rolls over to the next line and so on.

With VoIP did’s from same provider you can do it to…

Thanks. What do you mean next line? the next line is the destination # of caller?