Marlon Dutra wrote: > On 11/22/06, Xue Liangliang <[EMAIL PROTECTED]> 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. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
