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
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