[Asterisk-Users] Supervised transfer problem with BudgetTone

[The VoIP Connection]

Nicolas,

Just did some quick testing and the instructions are incorrect.  You need to
press "transfer" to complete the transfer instead of the second "flash".
This actually makes more sense.

Attended and regular transfer both work perfectly with the following
settings:

Enable Call Features: "Yes"
Disable call Waiting: "No"
Send Flash event: "No"

DTMF should be whatever * is set to, but in-band won't work properly if your
codec is anything other than U-Law.

By the way, the newest firmware also makes the long overdue conference
feature work properly.

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:611 at voiceserver.thevoipconnection.com
 

> -----Original Message-----
> From: Nicolas Schmerber [mailto:nicolas.schmerber at wanadoo.fr] 
> Sent: Thursday, August 11, 2005 10:41 AM
> To: asterisk-biz at thevoipconnection.com; Asterisk Users 
> Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Supervised transfer problem 
> with BudgetTone
> 
> The VoIP Connection a écrit :
> 
> >Section 4.3.7.2 from the Bugetone Manual:
> >
> >The user can transfer an active call to a third party with 
> announcement.
> >The user presses the “flash” button and hears a dial tone, then dial 
> >the 3rd party’s phone number followed by pressing send 
> button. If the 
> >call is answered, press “flash” to complete the transfer 
> operation, if 
> >the call is not answered, pressing “flash” button to resume the 
> >original call.
> >
> >Notes:
> >
> >• If attended Transfer fails, the BudgeTone phone will ring 
> the user to 
> >remind that another party is still on the call, the user can 
> then pick 
> >up the call using handset or speaker.
> >
> >Michael Crown
> >Managing Partner
> >www.thevoipconnection.com
> >321.989.6728 ext. 611
> >sip:611 at voiceserver.thevoipconnection.com
> > 
> >
> >  
> >
> >>-----Original Message-----
> >>From: Nicolas Schmerber [mailto:nicolas.schmerber at wanadoo.fr]
> >>Sent: Thursday, August 11, 2005 5:59 AM
> >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>Subject: Re: [Asterisk-Users] Supervised transfer problem with 
> >>BudgetTone
> >>
> >>steve at daviesfam.org a écrit :
> >>
> >>    
> >>
> >>>On Thu, 11 Aug 2005, Nicolas Schmerber wrote:
> >>>
> >>> 
> >>>
> >>>      
> >>>
> >>>>All the features I need work just not one : the supervised call 
> >>>>transfers. I know there are a lot of posts about that, but
> >>>>        
> >>>>
> >>none gave
> >>    
> >>
> >>>>me the correct answer (unless I missed it).
> >>>>   
> >>>>
> >>>>        
> >>>>
> >>>Hi,
> >>>
> >>>You'll need to switch to the CVS-HEAD version of Asterisk in
> >>>      
> >>>
> >>order to
> >>    
> >>
> >>>have supervised transfers.
> >>>
> >>>Steve
> >>>
> >>>_______________________________________________
> >>>Asterisk-Users mailing list
> >>>Asterisk-Users at lists.digium.com
> >>>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>To UNSUBSCRIBE or update options visit:
> >>>  http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>
> >>>
> >>> 
> >>>
> >>>      
> >>>
> >>When looking at a recent firmware changelog of Grandstream 
> , it says 
> >>BT should support supervised transfer, so shouldnt it work ?
> >>
> >>
> >>    
> >>
> >
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >  
> >
> Tried this manipulation a few minutes ago :
> 
> A calls B , B pushes flash button ( A is waiting with a mp3 
> played) B calls C pressing Send ; C answers B presses flash 
> button again ; C is so on hold (with a mp3 played) B hangs up 
> But A and C arent in connect ; the phoneof B rings ( to tell 
> someone is in wait : A)
> 
> So it seems to fail
> 
> What should i put in grandstream config for the next item :
> /Enable Call Features: Y/ N ?
> //Disable Call-Waiting: Y/N ?
> //Send DTMF: / in-audio / via RTP (RFC2833) / via SIP INFO 
> /Send Flash Event: Y / N ? / Any others Ideas ?.
> 
> Thx
> 
> Nicolas S.
>