On Nov 14, 2005, at 5:27 PM, C F wrote:
On 11/14/05, Leif Neland <[EMAIL PROTECTED]> wrote:
---- Original Message ----
From: "C F" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com> Sent: Monday, November 14, 2005
4:50
PM Subject: [Asterisk-Users] Problem with Cisco local conference and
hangup
Cisco 7960 gets a call from zap/1, hits conf to call out on zap/2,
then hits join, after a while cisco hangsup, at which point zap/1
and
zap/2 can still talk, shouldn't asterisk hangup on all three?
That is the way I would prefer it to work.
Like an attended transfer.
I cannot understand why, why not use attended transfer then?
Because the person performing the conference can speak to both
parties at the same time before hanging up and "transferring" the
call. With an attended transfer you can only speak to one caller or
the other, not both.
Also, it is worthy to note that this feature, as well as transfers to
outside numbers, creates a toll-fraud concern. Here's the scenario:
1.) Employee's wife calls employee at work - Local call.
2.) Employee conferences/transfers wife to inlaws in New Zealand -
Long distance call.
3.) Upshot: The company ends up paying the long distance for the wife
to speak to her mother.
(Of course, if your company is in NZ, this wouldn't be a big deal,
but it is for everyone else!)
So it should act differently if the outgoing call is internal or external.
Or outgoing calls should be blocked.
Or outgoing calls which are not tollfree should be blocked. (I have free
calls to all the subscribers of the same voisp as me)
Is there support in SIP for the voisp supplying the cost when setting up the
call?
Leif
_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@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
