Follow

FreePBX Configuration for OnSIP Trunking

Prerequisites

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:
    000 1 NPA-NXX-NXXX
    example: 000 1 212 555 5555
    where , 1 212 555 5555 is the outbound number you wish to dial
    One method of doing so involves setting the following

    Outbound Dial Prefix : 000


  • Prepend = 1
    Prefix = Leave blank
    Match Pattern = NXXNXXNXXXX
    Outbound Dial Prefix = 000

  • 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.

Was this article helpful?
0 out of 0 found this helpful
Have more questions? Submit a request

Comments

Powered by Zendesk