[Asterisk-Users] Features.conf - atxfer

[Dennis Walker]

On superviced you cancel on a zap channel you can cancel the transfer by
a hook flash this will send you back to the original caller.

On sip phones you hit the cancel button or if you have line buttons you just
pick the original callers line.

----------
From: 	Mike Holloway[SMTP:asterisk-users at citrini.com]
Reply To: 	Asterisk Users Mailing List - Non-Commercial Discussion
Sent: 	Tuesday, June 07, 2005 7:18 AM
To: 	Asterisk Users Mailing List - Non-Commercial Discussion
Subject: 	Re: [Asterisk-Users] Features.conf - atxfer


Reading through the code, I don't see a way of exiting the transfer and 
regaining the call with the customer, unless the third party hangs up or 
maybe doesn't answer and the dialplan doesn't do anything else with the 
call (send the call into voicemail).

I suggest you request this feature (http://bugs.digium.com), but as an 
interim solution you can create a dialplan for internal extensions that 
does not send the call to voicemail if unanswered, and only dials the 
third party for a limited amount of time (20 seconds?).

You could preface these special extensions with a sequence, such as 9, 
or 777 or whatever. Assuming your extensions are 1xx:

exten => _7771XX,1,Dial(SIP/${EXTEN:3},20)
exten => _7771XX,2,Hangup

-mike


Mark Johnson wrote:
> I am trying out the new atxfer feature from CVS-HEAD.  I set atxfer 
> equal to *7 and it seems to work OK.  I am having a problem getting it 
> to work the way a receptionist would want.  If an extension calls me, I 
> hit *7 and I hear the voice say "transfer".  I dial another extension.  
> If the newly dialed extension goes to voicemail, I can't figure out how 
> to get the original call back to tell them the person they are trying to 
> reach is unavailable.  Anything I try bridges the call and the caller go 
> into like the 2nd half of the voicemail greeting.  Is there some trick 
> to this?
> 
> Thanks!
> 
> Mark
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