Re: [asterisk-users] Agent Channel SIP transfer
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
Re: [asterisk-users] Agent Channel SIP transfer
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? -- MARLON DUTRA Propus GnuPG ID: 0x3E2060AC pgp.mit.edu http://www.propus.com.br/ http://hackers.propus.com.br/~marlon/ ___ --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
[asterisk-users] Agent Channel SIP transfer
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. -- Regards! Liangliang ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
