[Asterisk-Users] Call Transfers from SIP

[Dave Kitchen]

Sorry for such a basic question, but Googling and wiki searches haven't 
lead me anywhere.

Can a called phone transfer a call to another number?
In detail, I have ISDN/BRI via chan_capi -> * -> SIP to some xlite 
workstations.
All my dial() strings have tT on the end of them,  typical is:

exten => _${ISDN}${EXT_DAVE},1,wait(1)
exten => _${ISDN}${EXT_DAVE},2,wait(1)
exten => _${ISDN}${EXT_DAVE},3,wait(1)
exten => _${ISDN}${EXT_DAVE},4,SetCallerID(${CALLERIDNUM})
exten => _${ISDN}${EXT_DAVE},5,\
dial(Console/${RINGTONE_DAVE}&sip/${EXT_DAVE}&CAPI/264555:b${MOB_DAVE},${RINGTIME},,tT)
exten => _${ISDN}${EXT_DAVE},6,VoiceMail(u1${EXT_DAVE})
exten => _${ISDN}${EXT_DAVE},7,hangup
exten => _${ISDN}${EXT_DAVE},106,VoiceMail(b1${EXT_DAVE})
exten => _${ISDN}${EXT_DAVE},107,hangup
The Console call is just used to 'broadcast' a ringtone round the office.
What have I missed? I've tried xlite with forced inband DTMF and unforced,
I've tried to trace the source code of transfer actions, I'm lost!
Dave Kitchen