Attended/Consultation Transfer with SIP REFER/Replace: how?


I’m trying to implement a SIP attented transfer scenario as described in … ection-7.3.

I’m using 3 SIP phones, A, B and C. B is registered with Asterisk and acts as transferor. A and C are not registered with Asterisk and act as transferee and targed respectively. A calls B, B puts A on hold, then calls C. If C accepts the call, A should be transferred to C and B should hang-up.

I would expect that, after the trasfer, A and C are in direct communication, completely bypassing Asterisk (no active call legs between A and Asterisk and C and Asterisk), so I’m not consuming resources on the Asterisk box. Is that even possible? If so, does anyone know how?

Any help greatly appreciated.