Confirm that in Asterisk SIP Settings, External Address and Local Networks are set correctly.
I double checked and the IPv4 under Settings → Asterisk SIP Settings → NAT Settings is set correctly.
I also added all my local networks as well.
You could first try to get incoming calls working.
Incoming calls are working properly. 
If i try to make an outbound call on the other hand, this is what i get in realtime, when pjsip logging is enabled in the asterisk cli.
One thing i found out while enabling pjsip logging to the console was that occasionally a
SIP/2.0 405 Method Not Allowed appears.
If you still have trouble, post a screenshot of your trunk settings
(Sorry, using the link feature didn`t work for me like you said, even though i tried replacing the last dot
in the URL. I hope the way i did it is fine as well.)
This is the only official documentation for configuring SIP from my ISP i could find.
- SIP based VoiP RFC3261
- G711 alaw/ulaw codec
- DTMF transmission: RTP or InBand (RFC2833)
Thank you