We use conferences to allow a verto user to call and connect with an Asterisk channel. We would like to secure both signalling and media via TLS + SRTP, and I've read https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS a few times to understand how to do this. Note that that page has a broken link: https://wiki.freeswitch.org/wiki/Secure_RTP

First, is it still true that FS doesn't offer prebuilt installs (for Ubuntu) to support this kind of security?

Assuming that it must be compiled, I began to follow https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie with the additional first step of:  apt-get install libssl-dev