Connecting a SIP Trunk to a Remote SIP Extension

$WORK gives me a SIP extension on their Asterisk server for when I work from home.
I have an Asterisk + FreePBX box at home.
I wanted to be able to make/receive $WORK calls from home with my existing hard phones.
I didn't want to make any changes to $WORK's Asterisk server.

The SIP Extension at $WORK has the following settings:

          name: 1234
      callerid: Mick Pollard
   canreinvite: No
       context: default
      dtmfmode: rfc2833
          host: dynamic
      insecure: No
           nat: Yes
          port: 5060
       qualify: yes
        secret: 1234
          type: friend
      username: 1234

After some time researching it turns out this is not actually that hard.
The following is to be all done within FreePBX at home.

• Add a SIP trunk (use the details of your SIP extension on the office asterisk server)
• Add an outbound route
• add an inbound route

Add a SIP Trunk

The main difference here is you should leave "USER Context" & "USER Details" blank.

Add an outbound route:

The dial rules used here should be tuned to match the extension prefixes in use at your $WORK.
We have 4 digit extensions starting with either a 12 or a 22. I have also add a special prefix of 9|.
which allows me to route a call via $WORK. This is important so that clients get $WORK's callerID and not my home number !

Add an Inbound Route (optional)

I currently have an inbound route that allows any calls to go straight to a queue but you may want to change this.
You just need to create an inbound route that will match your WORK extension.

You should now be abe to make and receive work calls on your existing phones at home.