asterisk users - May 2005 - R: R: R: R: AT-320 + supervised transfer

Good...it almost works fine! I just have 't' in command Dial, but i also
have 'T'. Is it a problem ?
This is my Dial()

exten => 605,1,Dial(${GIORDANO NAT},60,Ttr)

I have only a problem: A and B are speaking, B calls C and ask it if wants speak
with A, C accept but if B hang up A is waiting and C get busy tone. To make it
works B don't have to hangup but habe to press forward button on the phone.
Can u check it ?


This is my features.conf (only featuremaps section):

[featuremap]
;blindxfer => ##                ; Blind transfer
disconnect => *0               ; Disconnect
automon => *1                  ; One Touch Record
atxfer => ##                   ; Attended transfer

Thanks for all
                     

-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill
Inviato: marted? 31 maggio 2005 16.21
A: asterisk-users@lists.digium.com
Oggetto: Re: R: R: R: [Asterisk-Users] AT-320 + supervised transfer

On Tuesday 31 May 2005 14:41, Giordano Grandis wrote:
> Hi Gavin,
> But...how atxfer work ?
Ehm, just the way I explained yesterday :) Just make sure you include the
't'
option to the Dial application, in the same way you need for the old-style
'#' blind-transfer to function.

gdh
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Hi Gavin,
I'm testing atxfer and it looks work fine, but i have a small problem:

A call B
B answer, dial atxfer extension and then the new peer (C)
If C does not answer the phone, A and B got hangup and cannot speak again

I set canreinvite to no.

Can u help me ?

Thanks
Giordano
                     

-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill
Inviato: marted? 31 maggio 2005 16.21
A: asterisk-users@lists.digium.com
Oggetto: Re: R: R: R: [Asterisk-Users] AT-320 + supervised transfer

On Tuesday 31 May 2005 14:41, Giordano Grandis wrote:
> Hi Gavin,
> But...how atxfer work ?
Ehm, just the way I explained yesterday :) Just make sure you include the
't'
option to the Dial application, in the same way you need for the old-style
'#' blind-transfer to function.

gdh
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Ok, just a thing...cuold is see a sample peer in tuou extensions.conf

Thanks
Giordano

-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill
Inviato: luned? 6 giugno 2005 21.50
A: asterisk-users@lists.digium.com
Oggetto: RE: R: R: R: [Asterisk-Users] AT-320 + supervised transfer

Hi again :)

I'm afraid I simply don't have any more suggestions... It 'works for
me' ...

 gdh

-----Original Message-----
   >From: "Giordano Grandis"<g.grandis@invidea.it>
   >Sent: 06/06/05 17:40:52
   >To: "Asterisk Users Mailing List - Non-Commercial
Discussion"<asterisk-users@lists.digium.com>
   >Subject: R: R: R: [Asterisk-Users] AT-320 + supervised transfer
     >Hi Gawin,
   >I'm newly testing the atxfer and i always the same question: if i
transfer a call to a peer that don't answer me, ho can i re-take the call.
   >Actually i got the call hanged up without the possibility the speack back
with my first caller.
   >
   >Thanks
   >Giordano                    
   >                     
   >


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