Call goes to hangup of transfer context

I am using attended call transfer with Asterisk 13 but whenever the transferer disconnects the call the hangup extension of transfer is called but in the asterisk 11 I have found that whenever transferer disconnects is redirected to hangup of the calling extension and when transferee disconnects is redirected to hangup extension of the transfer context.

I want to store which user left the conversation what time but this is causing me to update wrong data.
Please let me know if I am doing something wrong or there is other way to get it done.

Channel technology?

Features, register recall, or SIP native transfer?