Prerequisites

• FreePBX version 2.10 or newer is installed and running with appropriate permissions and behind a secure firewall
• Familiarity with configuring FreePBX and administrative access
• A valid OnSIP Hosted PBX account
• An OnSIP Trunking enabled user

Step 1: Gather information for the OnSIP Trunking User

You will need the following information from the OnSIP Trunking User in the Admin portal:

• Username
• Auth Username
• SIP Password
• Domain

You can find this information in the user detail pages under the Users tab in the Phone Configuration section.

Step 2: Add the OnSIP Trunking user as a SIP Trunk in FreePBX

Log in to the FreePBX Admin page

• Click on "Trunks", under the "Connectivity" drop down menu at the top
• Click on "Add SIP Trunk"
• Under the General Settings section
  Complete the following:
  
Trunk Name: OnSIP
Outbound CallerID: 15135555555
CID Options: "Force Trunk CID"

  The outbound "From:" section of an outbound SIP Invite request should look like this:
  From: "15135555555" <sip:hiro@example.onsip.com >;tag=as04cfd8df
  Where 15135555555 is your inbound DID.
  The "Force Trunk CID" option aids in ensuring that the Caller ID is configured for outbound calls to the PSTN.

• 

• Under the Dialed Number Manipulation Rules section
  It is important that all outbound SIP Invites should be of the format:
  1 NPA-NXX-NXXX
  example: 1 212 555 5555
  where , 1 212 555 5555 is the outbound number you wish to dial
  
  
• Under the Outgoing Settings section
  Complete the following:
  
Trunk Name: junction

  Peer Details
  
type=peer
host=sip.onsip.com
fromuser=hiro
fromdomain=example.onsip.com
username=example_hiro
secret=VPG3hockrifv
dtmfmode=RFC2833
insecure=invite
context=from-trunk (This context can be edited or omitted if you have a more specific inbound route, see your PBX documentation)


• 

• Under Incoming Settings section
  Leave this section blank (unless your specific design requires it)

• 

• Under the Registration section
  Complete the registration string:
  
hiro@example.onsip.com:VPG3hockrifv:example_hiro@sip.onsip.com
  generically:
username@domain:VoIP Password:authorized-username@sip.onsip.com


• 

• Click on "Submit Changes"
• Click on "Apply Config"

Step 3: Make test calls

Verify connectivity and correct signaling by placing test calls against a land line or cell phone.
You may need to make minor adjustments to your dialplan depending on your individual configuration.