asterisk users - Jan 2005 - New native assisted transfer (atxfer) usage info required

Hi, I would like to use the new atxfer (native assisted transfer, see 
mantis item #3241) , but I've partially been able to
make it work.
I can receive a call and then having the caller hear MOH while talking
with another extension (the one I want to transfer to), but then I can't 
make the caller and the trasferred talk hanging up or pressing any key
combination I'm aware of.
My features.conf are these ones:
[general]
language=it
parkext => 521                  ; What ext. to dial to park
parkpos => 522-525              ; What extensions to park calls on
context => parkedcalls          ; Which context parked calls are in
parkingtime => 120              ; Number of seconds a call can be parked for
                                  ; (default is 45 seconds)
;transferdigittimeout => 3      ; Number of seconds to wait between
digits when transfering a call
courtesytone = beep             ; Sound file to play to the parked caller
                                  ; when someone dials a parked call
xfersound = beep               ; to indicate an attended transfer is
complete
xferfailsound = beeperr        ; to indicate a failed transfer
;adsipark = yes                 ; if you want ADSI parking announcements
pickupexten = *8                ; Configure the pickup extension.
Default is *8

;featuredigittimeout = 500      ; Max time (ms) between digits for
                                  ; feature activation.  Default is 500

[featuremap]
; there are different from the default in features.conf.SAMPLE,
; and faster to type when apropriate, OMHO
blindxfer => #7                ; Blind transfer
disconnect => *0               ; Disconnect
automon => *1                  ; One Touch Record
atxfer => *7                   ; Attended transfer


And in the Dial() command I use the tT flag.
Is there any other special Dial() flag required in the dialplan?
If I hang-up the call is not transferred, nor if I presso *0 (disconnect).
I would also like to know how to go back to the caller, if the other
extension is  busy or doesn't answer or doesn't want to talk with the
caller.

Could someone provide me the exact settings required, and the keystrokes
needed to make it work (successful transfer and "aborted" transfer,
going back to the caller)? A sort of "atxfer for dummies" :)
I'm using the more recent CVS Head.
Btw, of course I know that I can have a "assisted transfer enabled"
SIP
phone, or use the 3 way calling of my TDM400, but I want to make this
feature of asterisk working without any client implementation (that is
the goal of atxfer).
Thanks a lot

Marco Menardi

I had it working.  My features.conf file is the same as yours except
for the [featuremap].  I use "##" for blindxfer and "**" for
atxfer.
In my dial plan I use "t" or "T" as the Dial() flag.

Make sure that you have beep.gsm and beeperr.gsm in the asterisk sound
file folder.  If these files are missing, transfer will fail.
> Is there any other special Dial() flag required in the dialplan?
> If I hang-up the call is not transferred, nor if I presso *0 (disconnect).
When I hang up the call is transferred.  When the transferee does not
want to answer, he/she hangs up and I am back with the original
caller.
> I would also like to know how to go back to the caller, if the other
> extension is  busy or doesn't answer or doesn't want to talk with
the
> caller.
My dialplan sends busy calls to voicemail, so that I can simply press
"#" to exit from the transferee's voicemail and get back to the
caller.

--JJL44


On Tue, 25 Jan 2005 20:37:49 +0100, Marco Menardi <mmenaz@mail.com>
wrote:
> Hi, I would like to use the new atxfer (native assisted transfer, see
> mantis item #3241) , but I've partially been able to
> make it work.
> I can receive a call and then having the caller hear MOH while talking
> with another extension (the one I want to transfer to), but then I
can't
> make the caller and the trasferred talk hanging up or pressing any key
> combination I'm aware of.
> My features.conf are these ones:
> [general]
> language=it
> parkext => 521                  ; What ext. to dial to park
> parkpos => 522-525              ; What extensions to park calls on
> context => parkedcalls          ; Which context parked calls are in
> parkingtime => 120              ; Number of seconds a call can be parked
for
>                                  ; (default is 45 seconds)
> ;transferdigittimeout => 3      ; Number of seconds to wait between
> digits when transfering a call
> courtesytone = beep             ; Sound file to play to the parked caller
>                                  ; when someone dials a parked call
> xfersound = beep               ; to indicate an attended transfer is
> complete
> xferfailsound = beeperr        ; to indicate a failed transfer
> ;adsipark = yes                 ; if you want ADSI parking announcements
> pickupexten = *8                ; Configure the pickup extension.
> Default is *8
> 
> ;featuredigittimeout = 500      ; Max time (ms) between digits for
>                                  ; feature activation.  Default is 500
> 
> [featuremap]
> ; there are different from the default in features.conf.SAMPLE,
> ; and faster to type when apropriate, OMHO
> blindxfer => #7                ; Blind transfer
> disconnect => *0               ; Disconnect
> automon => *1                  ; One Touch Record
> atxfer => *7                   ; Attended transfer
> 
> And in the Dial() command I use the tT flag.
> Is there any other special Dial() flag required in the dialplan?
> If I hang-up the call is not transferred, nor if I presso *0 (disconnect).
> I would also like to know how to go back to the caller, if the other
> extension is  busy or doesn't answer or doesn't want to talk with
the
> caller.
> 
> Could someone provide me the exact settings required, and the keystrokes
> needed to make it work (successful transfer and "aborted"
transfer,
> going back to the caller)? A sort of "atxfer for dummies" :)
> I'm using the more recent CVS Head.
> Btw, of course I know that I can have a "assisted transfer
enabled" SIP
> phone, or use the 3 way calling of my TDM400, but I want to make this
> feature of asterisk working without any client implementation (that is
> the goal of atxfer).
> Thanks a lot
> 
> Marco Menardi
> 
-- 
--JJL44

On Tue, 25 Jan 2005 19:33:00 -0500, Steven Frazier <lists@futuresync.com>
wrote:
> Could I ask you a question?  You don't have to flash to use the
transfer
> feature correct or do you?  I have tried it both ways and nothing happens.
No, I do not have to flash to use the transfer feature.  It works
pretty much like the old "#" transfer.  The sequence for blind
transfer is like this:

1. I press "##" keys quickly (two # keys within 0.5 seconds)
2. I hear the sound "Transfer", the other party hears hold music
3. I wait for dial tone.  After hearing the dial tone, I dial the
transferee.  I need to dial quickly because the timeout is set to 3
seconds.  After the last digit is dialed, I wait for the asterisk to
hang up and give me budy tone.  At this time I hang up.
4. The other party hears the ring tone for the transferee.  If the
transferee picks up, they can talk.

The sequence for attended transfer is like this:

1. I press "**" keys quickly (** is my atxfer key)
2. I hear the sound "Transfer", the other party hears hold music
3. I wait for dial tone.  After hearing the dial tone, I dial the
transferee.  I need to dial quickly because the timeout is set to 3
seconds.  After the last digit is dialed, I wait for 3 seconds and
then I hear the ring tone for the transferee.
4. If the transferee picks up, I talk with the transferee.
5. If I hang up, the transferee is connected with the caller.  If the
transferee hangs up, I am back with the original caller.
> I verified that I had beep.gsm and beeperr.gsm in my sounds directory and
> here is my features.conf file:
> 
> If I dial *1 (from a Zap or SIP) phone the person I am talking to just
hears
> the tones I press *2 or #1.  
When I press "##" (my blindxfer key) the other party does not hear any
DTMF tones.  If I press "#" (single #) the other party hears the #
tone after a short delay (I guess asterisk is delaying the tone to
make sure there is no more # coming).
> Does the automon work too?  Where does it put
> the file that it records at?
I have not tried this feature yet. 
> I made sure I have Tt in my dial command in my extensions.conf file as
well:
> 
> From my extensions.conf file:
> 
>   exten => 5801,1,Macro(stdexten,5801,SIP/5801,Ttr)
>   exten => 5802,1,Macro(stdexten,5802,SIP/5802,Ttr)
>   exten => 5803,1,Macro(stdexten,5803,SIP/5803,Ttr)
>   exten => 5804,1,Macro(stdexten,5804,SIP/5804,Ttr)
>   exten => 5805,1,Macro(stdexten,5804,SIP/5805,Ttr)
> 
>   exten => 5810,1,Macro(stdexten,5810,ZAP/1,Ttr)
> 
I think you can use only one transfer flag, either "T" or
"t" but not both "Tt".

Best regards,

--JJL44
> [general]
> parkext => 700                  ; What ext. to dial to park
> parkpos => 701-720              ; What extensions to park calls on
> context => parkedcalls          ; Which context parked calls are in
> parkingtime => 45               ; Number of seconds a call can be parked
for
> 
>                                ; (default is 45 seconds)
> transferdigittimeout => 3       ; Number of seconds to wait between
digits
> when transfering a call
> courtesytone = beep             ; Sound file to play to the parked caller
>                                ; when someone dials a parked call
> xfersound = beep                ; to indicate an attended transfer is
> complete
> xferfailsound = beeperr ; to indicate a failed transfer
> adsipark = yes                  ; if you want ADSI parking announcements
> pickupexten = *8                ; Configure the pickup extension.  Default
> is *8
> featuredigittimeout = 500       ; Max time (ms) between digits for
>                                ; feature activation.  Default is 500
> 
> [featuremap]
> blindxfer => #1         ; Blind transfer
> disconnect => *0                ; Disconnect
> automon => *1                   ; One Touch Record
> atxfer => *2                    ; Attended transfer
> [root@pbx05001 asterisk]#
> 
> Thanks for your help.
>