[Asterisk-Users] Agent channel X SIP Transfer on 1.2.9.1

[Leonardo Gomes Figueira]

Hi,

I wonder if on Asterisk 1.2.X calls from queue answered by Agent channel 
still must be transfered only by Asterisk internal transfer (features) 
like on 1.0.X ?

The wiki says on 
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Queue

"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."

But I did some tests and on 1.2.7.1 calls can be transfered via SIP 
transfers and the agents don't get locked. It's working fine with SIP 
but with IAX2 the agent channel gets locked until the transfered call is 
dropped.

On 1.2.9.1 the call is dropped after the transfer. If the call is 
transfered to a Polycom IP300 the caller is dropped too but the called 
(Polycom) is not dropped (but there is no audio of course) and if I run 
  a "show channels" on cli Asterisk segfaults.

I'm using 1.2.7.1 on production PBXs now cause 1.2.9.1 has bugs in 
queues/agents (I will post them soon) but what I really wanna know is 
what will be the standard to Agents transfer ? SIP transfers can be used 
or I must use attended transfer from features ? What about IAX2 transfers ?

  Leonardo