Keep in mind that with a SIP phone you are not communicating directly
with asterisk but with the phone which acts on your behalf with
asterisk. On traditional systems if you performed a hook flash to
transfer, you were definately signalling directly to the PBX. Now
when you push a button, hard or soft, on a SIP phone you are telling
the phone to perform as series of actions to accomplish a goal. It is
very much up to the phone software on exactly how the set behaves.
As stated previously, yes there should be a standard, but afaik there
are no standards bodies specifying the ui for voip devices.
On Apr 14, 2006, at 2:16 PM, John Novack wrote:
Jerry Jones wrote:
Yes it should all behave the way we are used to. However SIP IS
different. The exact behavior will be dependant upon the
individual hard phone.
Isn't that true only if it has a preprogrammed transfer key?
an Asterisk feature code should work as discussed.
There SHOULD be a way to make SIP phones work the same.
( easy to say, perhaps not so easy to do )
John Novack
This of course is if using SIP which we do not know yet...
On Apr 14, 2006, at 1:43 PM, John Novack wrote:
Michael Collins wrote:
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to
Person B
.) Person A hangs up the phone without waiting for Person B
taking the call
.) the caller get lost at this point !!
At this point the attended transfer should go into a blind
transfer.
The phone of Person B should still be ringing and the caller
shouldnt get lost.
I think this is the most usual behaviour of a call transfer
also on the cheapest systems on the market.
Could you remind us of what kinds of phones you are using, and
whether you're using SIP, Zap or something else?
Thanks!
-MC
I think the point of this post and other related ones is the
fact that there are attended and blind transfers, initiated by
different actions, where phone systems for at least the last 20
years have one action, or transfer.
The person initiating the transfer starts the procedure, and if
the destination extension answers, either through the facilities
of handsfree intercom or picking up the phone, the initiator and
the receiver can confer BEFORE the transfer is complete.
If, on the other hand the initiator either chooses to hang up
after starting the transfer, the transfer is then complete, and
the destination extension rings until answered or overflows into
voice mail.
In NO case should the call get lost. Attended and blind transfer
SHOULD start with the same action and be considered as ONE function
Irrelevant what phones are being used.
JMO
John Novack
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users