[Asterisk-Users] attended transfer issue

John Novack jnovack at stromberg-carlson.org
Fri Apr 14 12:16:17 MST 2006



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
>
>



More information about the asterisk-users mailing list