[Asterisk-Users] Switchboards

[brent clements]

On the actual switchboard side of this, how would asterisk handle
this...going to a central person/extension as well as notifying that
person who is calling and what line they are calling on.

I have a customer right now who is hesitant to move to asterisk
because of the switchbaord situation.

Right now they have an executone which as a few t1's coming in. DID's
are assigned to customers. In the executone switchboard application
when a call comes in on a DID, the called party's(one of the
businesses in the building)  name shows up on a console and the
receptionist can answer and forward the call as normal.


I guess it's as easy as all of the DID's point to a single extension,
then in that extension's context we could have an agi script which
looks up the name associated with the did and display it some how.

Something to think about...anyone has suggestions?



On 8/5/05, Tom Hayden <thayden at gmail.com> wrote:
> Well Don, it depends on how you get things setup from your telco. You
> could get a T1/PRI (or E1) and just trunk all your calls. Then you
> could have DIDs for each employee along with a primary number, which
> could receive multiple calls at a time (it's just a DID).  If you go
> with the POTS solution, then you'll need to get a channel bank and do
> things that way.  Frankly, it would be a hell-of-a-lot-easier and
> cheaper if you got trunked calls.  Why buy 23 POTS channels when you
> can probably get an equivalent number of channels + DIDs for much less
> money?
> 
> --
> Tom Hayden
> Astoria Telecom, LLC
> www.astoriatelecom.net
> 
> On 8/5/05, Don Brearley <donbrearley at hibbing.edu> wrote:
> >
> > Hello,
> >
> > I am still researching my dive into Asterisk at my workplace, and I was wondering about how switchboard
> > activities are handled..   Right now, a call comes into our switchboard, and the operator forwards them
> > to the appropriate line, thus freeing up the primary number and allowing more calls in.  Everyone on
> > campus has a direct-dial line as it is right now.  I want to eliminate most of those lines, and switch everyone
> > to extensions instead.
> >
> > If I understand correctly, with Asterisk, i'll need to figure out how many lines are in use at any time
> > (i'll say 20% to be safe) -- so I'll need to have roughly 25-30 POTS lines "on standby" for inbound calls?
> >
> > The way I see it in my head is, a call comes in on the primary number, and the operator will forward
> > them to the correct extension, and Asterisk will route the call to another circuit, freeing up the
> > primary line.  Is this correct?
> >
> > (Sorry for my lack of correct terminology.. still getting familiar!)
> >
> > - Don
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> --
> Tom
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>