[Asterisk-Users] queue caveats

[asterisk at anime.net]

According to http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue, under 
the "Notes" section:

"Transfers of calls that are answered out of a queue must be done using 
Asterisk '#' transfers (enabled with the 't' option above). SIP transfers 
result in the Agent remaining affiliated with the call until its eventual 
termination, preventing that agent from being offered another call."

Is this still true in asterisk 1.2.6?

-Dan