Should work if you followed that guide exactly.
Can you provide a sample call from the digium log?

Here is part of the log from the Digium, it shows the call as forbidden for some reason and congested. Let me know if you need more, I have to be missing something.
IP Addresses
Digium =192.xxx.xx.123
UCx = 192.xxx.xx.234

May 4 17:17:12 G100-59-ca-8e asterisk[29135]: VERBOSE[29240]: chan_sip.c:26214 in handle_request_do: --- (10 headers 0 lines) ---
May 4 17:17:12 G100-59-ca-8e asterisk[29135]: VERBOSE[29240]: chan_sip.c:4314 in send_request: Transmitting (no NAT) to 192.xxx.xx.234:5060: ACK sip:4591057@192.xxx.xx.234SIP/2.0^M Via: SIP/2.0/UDP 192.xxx.xx.123:5060;branch=z9hG4bK27c37218^M Max-Forwards: 70^M From: "N CAROLINA CALL" <sip:2523606252@192.xxx.xx.234>;tag=as61739bee^M To: <sip:4591057@192.xxx.xx.243>;tag=as2b2e31d3^M Contact: <sip:2523606252@192.xxx.xx.123:5060>^M Call-ID: 27686a8b7daf7d31103fe47d0900f11f@192.xxx.xx.234^M CSeq: 103 ACK^M User-Agent: Digium Gateway^M Content-Length: 0^M ^M ---
May 4 17:17:12 G100-59-ca-8e asterisk[29135]: WARNING[29240]: chan_sip.c:20969 in handle_response_invite: Received response: "Forbidden" from '"N CAROLINA CALL" <sip:2523606252@192.xxx.xx.234>;tag=as61739bee'
May 4 17:17:12 G100-59-ca-8e asterisk[29135]: VERBOSE[10752]: chan_sip.c:4027 in sip_scheddestroy: Scheduling destruction of SIP dialog '27686a8b7daf7d31103fe47d0900f11f@192.xxx.xx.234' in 32000 ms (Method: INVITE)
May 4 17:17:12 G100-59-ca-8e asterisk[29135]: VERBOSE[10752]: app_dial.c:1114 in wait_for_answer: == Everyone is busy/congested at this time (1:0/0/1)
May 4 17:17:12 G100-59-ca-8e asterisk[29135]: VERBOSE[10752]: pbx.c:4522 in pbx_extension_helper: -- Executing [s@sub-failover:11] NoOp("GTW/port1:1-0", "21") in new stack
May 4 17:17:12 G100-59-ca-8e asterisk[29135]: VERBOSE[10752]: pbx.c:4522 in pbx_extension_helper: -- Executing [s@sub-failover:12] GotoIf("GTW/port1:1-0", "0?evalstring") in new stack
May 4 17:17:12 G100-59-ca-8e asterisk[29135]: VERBOSE[10752]: pbx.c:4522 in pbx_extension_helper:

Forbidden is usually a wrong username or password.
In the pbx trunk settings -> outgoing -> peer details the defaultuser and secret matches what is in the gateway
In the pbx trunk settings -> incoming -> user context and details are left blank
In the pbx trunk settings -> registration string in the format of username:password@gatewayIP

Sorry theislandtech I made the changes yesterday and forgot what was what so here it is.

Ok if I set the SIP endpoint in the Gateway
ENDPOINT REGISTERS WITH GATEWAY and I set everything like above I'm unable to call in or out.

GATEWAY TO REGISTER WITH ENDPOINT and fill in the information needed I can call out but not in.

REGISTRATION TO NONE and fill in the information needed I can call out but not in.

UCx Trunk set as SIP like above OUTBOUND PEER info filled in but no INBOUND PEER information.

INBOUND route set to catch all DID’s and I’ve tried pointing it to an Extension, ring Groups and VM but all I get is reorder.

Time to start fresh. From the guide you referenced above

On the gateway create a new sip endpoint
1. enable advanced options no
2. give the endpoint a descriptive name
3. username anything as long as it matches whats in the pbx
4. make a strong password
5. for registration choose “Endpoint registers with this gateway”
6. use udp yes
7. use tcp no
8. use tls no
9. nat traversal yes (for the time being)

goto the gateway settings and config
1. click on call routing rules
2. click create call routing rule
inbound
a. give a descriptive name
b. call comes from - set to the t1 port
c. send call through - select pbx endpoint
outbound
a. give a descriptive name
b. call comes from - set to pbx endpoint
c. send call through - select t1 port
click save call rule

In pbx configuration
basic trunks
1. add sip trunk
2. trunk give a descriptive name
3. outgoing settings
a. trunk name (can use the name from above)
b. peer details enter the following:
host=(gateway ip address)
defaultuser=(username from gateway)
secret=(password from gateway)
type=friend
context=from-trunk
4. incoming settings
a. user context - leave blank
b. user details - delete contents if any and leave blank
5. registration
a. set registration string using the format (gateway username):(gateway password)@(gateway ip address)
6. save and apply

create inbound and outbound routes in the pbx as needed

Thanks theislandtech I started over with the same results but after a few changes I noticed on one of the calls in the UCx log the reject said “username and password” did not match. So after re-reading your instructions again I noticed your item “3a” under PBX Configuration and that is where the issue was I had a different name.
After renaming it calls started coming in.

I’m not really sure why it only affected Inbound and not any Outbound calls but it did and once it was changed they started working.

So now what I have is the Registration in the Digium set to None.
The UCx Peer Details under Inbound and the Registration string is blank and everything is working.

Do you think that’ll be ok or should I go back and change “Endpoint registers with this gateway”?

Thanks again for the help.

try setting it back to Endpoint registers with this gateway and the register string in the pbx.
You can change it back if it doesn't work.
I would watch the cdr reports for suspicious calls