3CX version 18.0.7 and later.
SMS requires a SMS enabled DID and a 10DLC form
You can restrict port 5060 UDP (SIP signaling used for call setup and call teardown) to:
However, since we RELEASE THE MEDIA of the actual call to one of our 20+ carrier's media gateways, (and they have hundreds of gateways around the country), you really don't know where the RTP media will be coming from. That is why you must allow all traffic on 10000-20000 UDP (or the set UDP range of your PBX), or else you may get one-way audio.
⦁ Navigate to the “SIP Trunks” menu on the left menu column and click "Add SIP Trunk." Select “Generic” as the Country and “Generic SIp Trunk” as the Provider. Enter your Trunk number in the “Main Trunk Number” field. Click OK.
⦁ On the “General” tab of the Trunk configuration, your “Registrar/Server/Gateway” host should be gw.siptrunk.com with the “Auto Discovery” box checked. Outbound prxoy should be gw1.siptrunk.com.Below in the “Authentication” section, enter your Trunk Number as the “Authentication ID” and your “Trunk Password” (found in the SIP.US Customer Portal under SIP Trunking --> SIP Trunks) as the “Authentication Password.”
⦁ Select the “Options” tab in the Trunk configuration. Scroll down to “Advanced” and confirm the template has “Alternative Proxy” configured with “gw2.siptrunk.com” in the field. This will enable failover to our redundant gateway if the zombie apocalypse begins in Atlanta and gw1.siptrunk.com is taken down.
⦁ Select the “Inbound Parameters” tab in the Trunk configuration. Change the “CalledNum” parameter from “To: User Part” to “Request-Line URI: User Part.” Click OK to save the Trunk configuration.
For general dialing of 11-digit North American numbers, this route will take the number and route it directly to SIPTRUNK.
For general dialing of 10-digit North American telephone numbers, this route will route dialed numbers with a length of 10 digits and add a "1" to the front of the number. Then it will route to SIPTRUNK. Pay special attention to the "Prepend" column.
For Emergency Calling, this route will take all calls that begin with a "9" and are 3-digits in length. These calls will then be routed to SIPTRUNK.
For International Calling, this route will take all calls that begin with "011" which is the International Exit code, and then send the calls to SIPTRUNK. after stripping the first digits (011) from the number. Specifically, notice the "3" under the "Strip Digits" column.
For inbound call routing, you will first need to obtain DID (Direct Inward Dial) numbers from SIP.US. This can be done in your Customer Portal under SIP Trunking --> Order Telephone Numbers.
Once you have at least one DID, navigate to “SIP Trunks” in your 3CX. Edit your existing SIPTRUNK trunk and select the “DIDs” tab. Click “Add Single DID” and enter your DID in an 11-Digit format (i.e. 14045964200) and press ENTER. Click OK to save changes.
⦁ Navigate to the “Inbound Rules” menu. Click the “Add DID Rule” button. Name the rule whatever you'd like, and verify that the correct DID is selected in the “DID/DDI” drop down. You may then use the “Route calls to” section to route inbound calls accordingly.
Now that the Generic trunk has been created. Go to the newly created SIP trunk and over to the SMS Tab. Copy the Webhook URL.
In the SIPTRUNK portal, go to MESSAGING and click on the Webhooks. Create new Webhook. Enter the copyed Webhook URL. Enter a description and click create. The Webhook is now created.
Go to API KEYS. Create new API KEY. Enter a description and click create.
Make sure to save the Secret Key for any future use.
Copy the 3CX API Token
Go back over to the 3CX SMS tab. Enter the copied 3CX API Token here:
Enter https://messaging.siptrunk.com/3cx/ the Provider URL then press OK.