If not, what does the text represent and what's the correct way with, effectively a duplicate trunk with different credentials.
All of this would make a lot more sense after sitting down for an afternoon and reading about how SIP works, and then experimenting on sandbox servers. I don't wish to be condescending. The questions in this thread just don't make a lot of sense in terms of how SIP actually works.
For outgoing calls, you are sending the call request from your PBX to Flowroute with some credentials to tie the call to a known account.
For incoming calls, all you need is a list of IP addresses that are allowed to send calls to you. They do not authenticate to you in any other way than being a known IP.
If you're doing this with PJSIP, you put the list of known IP addresses, or a subnet definition, in the Match field.
If you're doing it with chan_sip, you define a peer for every IP address. The syntax of [flowroute-ip1](flowroute) tells Asterisk to copy the peer that was previously defined named "flowroute" (this is what you named it in the FreePBX chan_sip trunk field called Outgoing - Trunk Name) and use all the same settings for it except for whatever follows (host= ).