[Asterisk-Dev] bugs #2700 and #2721 (chan_agent transfers)

[Kevin Blackham]

The problem with the agent channel not being freed, or callers being
disconnected on native SIP transfers (where # works fine, see mantis
bug IDs in subject) are plaguing me and at least two other call center
implementors.  Myself and one other (twilson) have come up with dial
plan hacks to avoid chan_agent, but duct tape sucks.  Our workarounds
are nearly identical, and can pretty much get the functionality
reproduced, with the exception of wrapuptime.

This has pretty much been the only problem keeping me from rolling *
company-wide.  What is the status of this bug slash design issue being
addressed?  My preference, and what seems like the best idea, is to
move some the agent tracking code into app_queue, and get rid of
chan_agent altogether, which I can dedicate some time to. 
Alternately, to have chan_agent handle SIP transfers in the same way #
transfers are.  I don't understand channels enough to know what that
involves.  Having looked at chan_agent.c and receiving an immediate
headache and nausea, I have no idea where to start on that part.