SIP trunk is just a name that people put on a a particular way of using SIP to pass routing digits to a remote location. Whilst it is an easy construct to create in Asterisk, it is not something that Asterisk actually knows about . You can use similar constructs using many of the other technologies supported by Asterisk. E.g., before SIP you might have used dahdi and ISDN, and, if you expect to handle more than one erlang of traffic (more than one call at a time), IAX can allow you to use less bandwidth.
Asterisk imposes no restrictions on the remote breakout requested here, but again it doesn’t have any special knowledge of this.
Basically you use any channel technology that is capable of passing digits to the remote site, and at the remote site you handle it exactly like you would handle a local device with the caveat that you cannot authenticate it based on caller ID if you also want to pass the caller ID through.
Your source dialplan needs to recognize numbers subject to remote breakout and forward them in a similar way to that in which you would do the PSTN or an ITSP.
In fact, I think some smaller ITSPs use Asterisk, and all ITSPs are basically operating remote breakout configurations.