Video 2: How to Set-up SIP Trunking Service
This video instructs you how to setup your SIP.us account and free SIP trunk on your new PBX.
1. Browse to your DigitalOcean account > Look for your droplet’s IP Address in your browser
2. Open new browser > Add “admin” and the droplet’s IP address by typing it into your browser: “IP address/admin”. This is how you’ll log into your server.
- Enter “maint” for the “User name”
- Enter “Password” which is your previously generated password
- Everything on your FreePBX screen should show green
3. Get a free demo account and demo SIP trunk from SIP.us > Select “Get Started”
- In this FreePBX template we have included a module for both SIP.us and SIPTRUNK.com service which generates a setup key to automatically generate and setup a SIP trunk
- For this example we’re going to use SIP.us as our SIP trunking provider
- The SIP.us demo is good for 14 days and up to 60 minutes of calling
- Enter your first name, last name, and email after selecting “Get Started”
- “Accept” the terms of service
- Click “confirm email”
- Confirm new password
- Skip, or watch the tutorial
- You will now have a SIP.us demo account number, and SIP.us customer number.
4. From your SIP.us demo account > Select “Trunks”
- Your trunk is already fully working
- Click “Get a Test Number” (you can also easily convert the demo trunk into a paid trunk here)
- Click the “FreePBX Config” tab, and click the link labeled ‘HERE’, to view your secret key
- Copy the secret key
5. Browse to FreePBX admin screen > Select “Connectivity” drop down > Select the “SIP.us” module
- Add your SIP.us account by pasting the secret key
6. On your FreePBX server click your account number (right side of page) to setup dialing through your SIP.us demo trunk
- Click “Refresh Remote SIP.us Settings”
- Click the checkbox to “Enable SIP.US Trunks”
- Click the checkbox to “Enable SIP.US Outbound Routes”
- Click “Save”
- Click “Apply Config”
7. Select “Reports” drop down menu > Select “Asterisk Info” module
- Click “Registries” (right hand side of the page)
- You’ll see your 2 SIP registrations to both gw2.sip.us and gw1.sip.us
8. You can also check to see registrations though the DigitalOcean console
- Click “Console Access”
- Type “asterisk -r” to bring up the command line
- Type “sip show registry” and you should see the same two lines that appeared in step 8
- Type “sip show peers” this will show that you’re pinging the SIP.us gateways, the latency, and that you’re using 5060.
- Everything should show “OK”
You should now have your trunk setup correctly (and very quickly thanks to the SIP Trunk module). If any of the registration checks say that it’s not “OK”, you’ll need to go back and fix one of the steps.
Why use SIP.us?
The FreePBX template we use for DIY PBX integrates SIP.us SIP Trunking service, making it far easier to configure your PBX. SIP.us provides highly secure, month to month SIP Trunking at very affordable prices and an intuitive management interface.
What are SIP Trunks?
SIP trunks are essentially a telephone line. A SIP trunk contains 3 primary components: channels (concurrent call paths), DIDs (phone numbers), and minutes (either domestic or international (note the idea of local and interstate long distance in the North America has disappeared)).
You can port your analog or PSTN numbers (DIDs) to a SIP trunking provider in almost all cases. You can use SIP Trunking service on any current IP/Hosted/Virtual PBX including your DIY PBX.
Why use SIP Trunks?
SIP Trunks offer lower dial tone costs, better and more flexible services, innovation, mobility, and lower infrastructure costs.
Why move from the PSTN to SIP Trunking?
The PSTN is coming to an end. Just like all technology, telephony is rapidly moving into a more flexible and efficient way of providing telecommunications using SIP Trunks. Most major telco networks have stopped offering customers basic analog services as early as 2007, and by 2019 the move away from providing analog service will in most cases be completed.
How long is the Free Trial for the SIP.us demo?
14 days, and up to 60 minutes of calling.