In that case, you should be using chan_pjsip, not the deprecated and unsupported chan_sip.
The address in zoiper should be the address of Asterisk as seen by zoiper, which would normally be the LAN address, or even 127.0.0.1, if is on the same machine. I would advise against starting with it on the same machine, as that introduce additional complications.
You should start with echo and playback applications, rather than going straight to an outgoing call.
Only use type=friend if you have more than one phone with the same address (but chan_pjsip is rather different, so don’t bother learning chan_sip).
The register line looks incomplete.
defaultuser is not useful for a non-dynamic peer.
username is the obsolete name for defaultuser, is rarely useful, and then only with a default domain.
It is not clear why nat= is not left at default, although it would be unusual for it to cause any harm.
Just FYI, I have been successfully doing some SIP testing using Twinkle as the SIP-phone client, connecting to an Asterisk server on the same machine. I keep them out of each other’s hair by letting Twinkle use port 5060, and putting Asterisk on another port, e.g. 506.
This way I make my embarrassing mistakes on my own private LAN.