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IMPORTANT: If you are using FreePBX version 16 or higher, you need to go to Settings > Asterisk Settings, find the SIP Channel Driver setting, and set it to "Both". Then create a SIP (chan_sip) trunk.
How to setup SIP Trunk in FreePBX
1. Log into the web interface by typing the IP address of your PBX in a web browser. You can obtain this address from your router's DHCP client list.
Go to the PBX top menu, then on the left menu click Trunks, and "Add SIP (chan_sip) Trunk" link on that page. (If you only have pjsip, see Important Note at top of this page.)
Construct the SIP Trunk as follows...
Trunk Name: 10digitnumber (this is your CLONE LINE number)
Outbound Caller ID: <10digitnumber> (as above, but with < and > around it)
In the Dial Pattern Manipulation Rules, enter 1 in the prepend field, and NXXNXXXXXX in the Match Pattern field.
In Outgoing Settings, enter Trunk Name again: 10digitnumber (same as above Trunk Name)
In Peer Details, paste the following, then replace the CLONE LINE number, password and server fields with info sent separately.
username=[CLONE LINE Number]
type=peer
secret=[CLONE LINE SIP Password]
qualify=2000
nat=yes
insecure=port,invite
host=[CLONE LINE SIP Server]
dtmfmode=auto
fromuser=[CLONE LINE Number]
context=from-trunk
disallow=all
canreinvite=no
allow=ulaw
Leave Incoming Settings blank.
And finally, build the Register String as follows:
CLONELINENUMBER:CLONELINEPASSWORD@CLONELINESERVER:PORT/CLONELINENUMBER
Submit changes, then apply changes to activate the trunk.
2. Create a generic inbound route, outbound route, and any extensions, menus, etc, that you like.
Go to Inbound Routes and add a new route, called Default or whatever you like. Leave everything set to default, except the last field, select a default destination for your calls, such as an IVR, Ring Group or Extension.
Save changes.
Go to Outbound Routes and add a new route, we usually name it after the trunk phone number.
Enter <phonenumber> (using <>) for the caller id field.
Use the Dial Pattern wizard and select 7/10 digits, long distance, emergency and information. We do not recommend adding an international pattern unless you plan to make international calls.
Select one or more trunks in order of preference to be used for this route.
Save and apply all changes.
3. Proceed with adding extensions, ring groups, menus, etc.
Remember, with CLONE LINE, there's no "per extension" charge. Create as many extensions as you like.
IMPORTANT NOTE: If you complete the above, and your inbound calls are not arriving, you likely need to setup the "external ip (externip) setting on your pbx. Go to Settings > Asterisk SIP Settings and click the "Detect Network" button. That usually fills everything for you automatically. If not, you may have a nonstandard network, such as "double NAT", or an odd DHCP setup, or maybe some kind of address conflicts that need to be resolved.
