What are SIP trunks. There is no such concept in SIP or Asterisk.
Asterisk supports REFER and REFER/Replaces when acting as a UAS, whether the peer is a phone or another PABX.
It also supports REFER, but not REFER/Replaces, as a UAC. Again it doesn’t care whether the peer is a phone or another PABX.
This is for chan_sip, but I assume that the same is true for PJSIP.
Asterisk doesn’t automatically de-trombone calls. If it receives REFER when acting as a UAS, it will not forward REFER upstream, but will rather just brdige the call internally. You have to use the Transfer application, rather than DIal, to send a REFER.