I think this has been fully discussed on the FreeSWITCH list. Its really a mechanism of the phone/UA and how it bypasses media to handle attended transfers.
In your use case the attended transfer is to a FreeSWITCH media server. I think this is an issue in the sip stack and is found in Cisco phones as well as some grandstreams too. I don't think attended transfer will work on Cisco's (from sip 8.2 to 9.3). There is noone updating the Cisco plugin or maintaining it, cited due to too many records route issues with each and every sip release lately. If it were me, and its not, I'd get Poly com phones. On Aug 1, 2012 9:09 PM, "Tony Graziano" <tgrazi...@myitdepartment.net> wrote: -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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