Call drops after attended transfer

When user A try to transfer call to user B, they both hung up. In console I see
WARNING[xxxx]: file.c:766 ast_readaudio_callback: Failed to write frame
WARNING[xxxx]: features.c:2591 builtin_atxfer: Failed to play transfer sound!

Asterisk is 10.3.0 from freepbx distro. I found this -
As I understand, I need to upgrade * to version 10.5.0.
So my question is - what should I save in my system, when I compiled and reinstall asterisk from source? Freepbx web interface will work with new version without reinstall?
Sorry for poor english