Some of those settings changes require that Asterisk be restarted (not just reloaded). You also may have to reboot the endpoint after restarting Asterisk. If you still have trouble with this, please post details.
I’m fairly sure that this is the same issue as:
https://community.freepbx.org/t/migrating-sip-to-pjsip-trunk-problem-incoming-call-drops-after-32-seconds/
The one in the current thread is a little more complex because the failure is source carrier dependent, but I suspect that’s because a different Vodafone server is involved in the two cases.
The failure is caused by pjsip sending the 200 OK response to an incoming INVITE with a Contact header that does not contain a user part. I believe that is in conformance with the RFCs but differs from industry practice. It appears that some trunking providers fail to parse the header correctly and therefore don’t send an ACK to the correct address. When I looked at the issue in 2018, pjsip had no setting that would force a user in the Contact field (in this situation).
It’s possible that pjsip has since been updated to provide this option, though if so there is nothing in the FreePBX GUI to enable it.
Though IMO this is technically a problem on the trunking provider side, we are very unlikely to get them to correct it, so we need the fix in pjsip.