Hi, I'm having problems with sip remote attended transfer using 2 asterisk boxes (same version, latest 1.4.X). Whenever I transfer from a call from box A to a call on box B, one call leg of the transferring phone is not disconnected (the one that is normally dropped by server side, phone disconnects the other one). The same situation works perfectly with local attended transfer.
Is anyone using remote attended transfers with asterisk? Does it work for you? Do you use any workarounds? I'm asking here, because it would be strange if that functionality was broken since 1.4.8 and noone noticed ;) Exact scenario I'm using is described in the bug: https://issues.asterisk.org/view.php?id=15833 Thanks for any help. -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: [email protected] | www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
