We are having problems getting call transfers to work with polycom 301 sip phones. We call from the pstn to the polycom, and have the polycom do either a normal or blind transfer to another sip phone or a voice mail extension. Trying to dial the polycom after the transfer is completed only gives a busy signal, We have to restart asterisk when this happens to make the phone reachable again. We tried this on a fresh version of Elastix and are using the polycom 2.1.2 firmware. Our other sip phones are able to transfer calls without this issue. The polycom phones are able to transfer calls correctly with the '##' feature code, but we would like to use the transfer method built into the polycomes. It looks like it may be a bug (http://track.sipfoundry.org/browse/XPB-396) with polycom phones. But we are curious if anyone knows of a workaround for this problem?