By default, Freepbx has pjsip Port to Listen On set to 5060 and chan_sip Bind Port set to 5160.
Set up a pjsip trunk (you shouldn’t be using chan_sip at all) with Registration None and Authentication None. Set SIP Server to 190.64.60.17 . If Antel will send calls from multiple addresses, set Match (Permit) to a list of those.
Asterisk should then respond to OPTIONS with a 200.