[asterisk-users] Agent Channel SIP transfer

[Russell Bryant]

Marlon Dutra wrote:
> On 11/22/06, Xue Liangliang <xueliangliang at gmail.com> wrote:
>> Hi, we are using asterisk 1.2.13. When callbacklogin agent transfer
>> call using SIP phone's transfer feature, he is always in busy status
>> and cannot answer any more incoming call from queue until the
>> transferee hang up the call.
> 
> I'm experiencing the same problem here with Asterisk 1.4.5.
> 
> Is there a solution for that?

I would encourage you to check the bug tracker for reports of this.  I think it 
may already be there but I can't remember.  If it's not, feel free to report it 
and we'll work on it getting it fixed.

On a related note, most people have reported much more stable results when 
building their own callback login system using dialplan logic and dynamic queue 
members.  We even posted a document to provide some examples for how to do it.

http://svn.digium.com/view/asterisk/branches/1.4/doc/queues-with-callback-members.txt?view=co

-- 
Russell Bryant
Software Engineer
Digium, Inc.