SIP reinvite so IAX doesnt get used twice

Hi There I have currently have incoming calls setup like this
Analog PSTN -> Asterisk -> IAX to another Asterisk -> reception

The receptionist usually transfers these calls back to the the originall Asterisk box that they came from but this takes up and IAX channel there and back.

Is there a way to use sip reinvite to that the IAX channels are taken all together?

Thanks,
Justin