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.
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