[Asterisk-Users] Supervised transfer problem with BudgetTone

[Nicolas Schmerber]

Hi all,

I'm quite new on this mailing list, and I discover the asterisk world.
I m experimenting a PBX with SIP phones, grandstream budgetone (not 
expensive for tests)

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).

Here is my config :

2 sip phones BT102 with firmware : 1.0.6.7 (the last at this day)
Asterisk from debian stable (1.0.7)
a Bri connection arives on the PBX, this works , and can reach the sip 
phones; but the call transfer only works in blind ( no ability to speak 
to the transfee to introduce the incoming call).
Here are config files :

extensions.conf :

NICO = SIP/nico
CEDRIC = SIP/cedric

[default]
include => incoming
exten => 22,1,Dial(${CEDRIC},20)
exten => 23,2,Dial(${NICO},20)

[incoming] ; the BRI stuff
exten => 9692,1,Dial(${CEDRIC},20)   ; if numerber arriving on bri 
finishes by 9692 dial Cedric
exten => _969X,1,Dial(${NICO},20)      ; else dial Nico


features. conf :

[general]
atxfer => *5

So I d like to know the params for the BT phones, the asterisk config , 
and the procedure ( for example should i press *5 when i want to release 
the line and etablish caller => transfee ) and so on .

Thanks in advance