[Asterisk-Users] Supervised transfer problem with BudgetTone

[Eric Wieling aka ManxPower]

Olle E. Johansson wrote:

> CVS head of Asterisk supports attended transfers native in Asterisk, not
> really SIP attended transfers. Work is in progress in that area, but
> will require quite a lot of changes to the SIP channel so I am not sure
> whether we will be able to support it in 1.2 or not. Definitely in the
> 1.4 release.

What is the specific problem?  We hav been doing supervised transfers 
with 1.0.x and Polycom phones for several months.

-- 
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. "r" makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.