FreePBX 14 – Trunks, Extensions and Routes

Now that we have networking setup, we can move onto setting up the rest of FreePBX.  Let’s start by setting up a trunk to our SIP provider:

Trunks

A trunk is a connection to another PBX, in VOIP-Speak.  This is a slightly different definition than when trunking is in Network-Speak, as many of the Telephony technologies are similar but most always different.

1.

Log into your PBX.

2.

Under Connectivity / Trunks, click +Add Trunk and select +Add SIP (chan_sip) Trunk:

  • Trunk Name: Give this a name you can remember that is relevant to the service.
  • Outbound Caller ID:  Put in your phone number from the SIP provider here, if you have purchased DIDs.
  • Click the Sip Settings tab when done.

3.

In the sip Settings tab:

  • Trunk Name: Give it the same name as you did before.
  • Peer Details: Replace everything here with this text:

type=friend
secret=<the password your SIP provider gave you>
username=<the username your SIP provider gave you>
host=<the service address that your SIP provider gave you>
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
insecure=port,invite
fromdomain=<Your SIP provider’s address>

  • Click the Incoming tab next:

4.

Incoming sip Settings:

  • USER Details: Clear out any existing text.
  • Register String:  Put in <SIP Username>:<SIP Password>@<your SIP Provider.com>  These values were used above.
  • Click Submit when done.

5.

Click Apply Config at the top right:

6.

Now go to Reports and select Asterisk Info.

7.

Select Chan_Sip Info at the right and see if your trunk has Registered with the SIP provider:

Extensions

Now that we have our trunk set, let’s setup an extension.  An extension is an endpoint to where a call will “terminate” to, as well as a phone will register to.

1.

Log into your PBX.

2.

Go to Applications / Extensions in the PBX.

3.

Click + Add Extension and select + Add New Chan_Sip Extension:

  • Extension: Choose an extension number
  • Display Name: Enter the user of this extension (or the purpose such as “Front Desk” or “Fax”)
  • Outbound CID: Enter the Phone Number you got with your SIP Provider.
  • Secret:  This is randomly generated, I enter something easier to remember.  This will be used as the password to register phones to this extension.
  • Password For New User:  This is also randomly generated, leave this for now.
  • Click Submit when done.

4.

Add more extensions as needed.

5.

Click Apply Config at the top right when you are done:

6.

Now click Connectivity / Firewall and choose the Network tab:

  • Enter new IP or Hostname here:  Enter your LAN network IP here.
  • Select Trusted (Excluded from Firewall)
  • Click Save when done.

7.

Finally, go to Reports / Asterisk Info and click Chan_Sip Info:

  • Under Status in Chan_Sip Peers, all your extensions should show up as OK if already registered to a SIP phone.

Inbound Routes

To get a SIP Provider’s phone number to ring a phone, we’ll need to create an Inbound Route.

1.

Log into your PBX.

2.

Go to Connectivity / Inbound Routes.

3.

Click +Add Inbound Route:

  • Description: I put in the SIP provider and the DID here
  • DID Number: Enter the phone number your SIP Provider gave you.
  • Set Destination: Choose Extensions and select the extension that will ring when this number is dialed.
  • Click Submit when done.

Outbound Routes

To place a call that is not local to the PBX (an extension), we’ll have to make an outbound route to the SIP Provider.

1.

Log into your PBX.

2.

Click Connectivity / Outbound Routes

3.

Click + Add Outbound Route:

  • Route Name: Enter a route name
  • Trunk Sequence Matched Routes: Select the trunk you made in the previous steps.
  • Click Dial Patterns tab when done.

4.

Click the “Dial patterns wizards” button and select all Digit Pattern and US buttons:

  • Click all the Digit Patterns and US / Long Distance buttons.
  • Click Generate Routes when done.

5.

You should see all the possible outgoing routes now, click Submit:

6.

Click Apply Config at the top right when you are done:

Now you should be able to receive and place calls using your SIP Provider.

That completes this tutorial.

One thought on “FreePBX 14 – Trunks, Extensions and Routes

Leave a Reply

Your email address will not be published. Required fields are marked *