asterisk users - Jun 2009 - the correct way to setup a transfer with REFER in SIP

Hi to all

excuse me but i don't understand what is the correct configuration
needed to setup a transfer with REFER in SIP.

I've tried the transfer() application, but i've experienced some
problem, i can't reproduce the error in a clear debug environment but
randomly the call crash before to be transferred to the final peer.
on the wiki (http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer)
it is reported as a partial implementation of the REFER functionality.

I've tried both atxfer and blindxfer in features.conf but it seems
that asterisk make a simple Dial between the two peers.

I've also taked a look at
ChannelRedirect(channel|[[context|]extension|]priority)  but it
doesn't seem to be what i need.

This is my scenario:

A is a SIP Phone registered on the SIP PBX "test"
B is a SIP Phone registered on the SIP PBX "test"

Asterisk is registered on the SIP PBX "test" with the user C

D is a SIP Phone registered on Asterisk.

1) A dial C
2) C (that is Asterisk) execute the dialpan and dial D
3) A and D talks directly as the native bridging is enabled by
canreinvite=yes and the codecs are compatible
4) D transfer the call to B

What is the configuration needed for the 4th action?
My aim is to make a REFER to B at test and free completely Asterisk.

Thanks to all in advance, bye.

-- 
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nik600
http://www.kumbe.it