I do not understand this part:

Three pieces of information are required to add a SIP URI forward from OpenSIPS to your Asterisk server using the AVP asterisk-add-forward script:

UUID of SIP URI (from any SIP phone, dial UUID@opensips.yourdomain.com to connect)

When I call: UUID@opensips.mysipserver.com from linphone on my pc, it does nothing but "Abort".

Any suggestions?

[WM: Lots of moving parts to this. Does the Asterisk CLI show the incoming call at all? If not, I’d temporarily disable the firewall and try the call again. If you still don’t see it on the CLI, then reenable the firewall and the issue is on the OpenSIPS server or your softphone registration to it. Try another kind of softphone and see if it registers. If so, see if you get the same failures. If you do, then the problem lies with your forwarding on the OpenSIPS server. Can you forward anything to your Asterisk server? A good place to start would be the news or weather apps which don’t require a registration at the incoming end. Finally, what version of the installer does pbxstatus show? If you’re below 1.2, you probably should reinstall and try again.

For further troubleshooting, you’ll need to open a thread on the PIAF Forum. It’s not practical in blog comments. Sorry.]