Configure a handSIP VoIP SIP Trunk in 3CX PBX
Requirements for using a handSIP VoIP SIP Trunk with 3CX
Per 3CX official documentation, the following requirements should be met in order to use 3CX with handSIP.
- A firewall/router/NAT device that supports STATIC PORT MAPPINGS. Often routers will perform port address translation, which will cause problems such as one way audio, failing inbound calls, etc. It is also highly recommended that you have an FQDN that resolves to a static external IP. If your external IP changes intermittently, inbound calls will fail. See the Firewall & Router Configuration for details on configuring your firewall/router/NAT device.
- Adequate Bandwidth. VoIP is real time, and as a result, it places a demand on your Internet connection. As a rule of thumb, each call will consume approximately 30-120 kb per second, depending on which codec you use.
Requirements for VoIP Gateways
If you plan to use a VoIP gateway with your 3CX/handSIP configuration, the following requirements should be met.
- A 3CX supported VoIP gateway. Supported gateways have been tested by 3CX and are automatically configured with their correct settings. If using the default configuration, 3CX will also provide first line support of their use with 3CX Phone System. A list of the latest supported gateway hardware, can be found on the Supported VoIP Gateways & ATA’s page.
Step 1: Create a handSIP VoIP SIP Trunk Account
handSIP has been tested and certified by 3CX. In order to make and receive calls over handSIP using your 3CX, you must first have handSIP account created. If you are an existing handSIP customer, or have a free trial account, you will need the account credentials provided to you by your Imecom representative. If you do not have an account, you can:
- Request a Free Trial Account or
- Sign up for a Paid Account
Step 2: Conduct the Firewall Test
3CX will prompt you to conduct a Firewall Test. It is common that the internet facing firewall sitting between 3CX PBX and handSIP is not configured properly or is not able to correctly route VoIP traffic. To check the firewall configuration, it is important to perform a firewall check using the built-in firewall checker. To do this:
- In the 3CX Management Console, go to the System Status page.
- In the section PBX Status select the Firewall Check entry.
- Click Run.
- Ensure that the tests for the SIP Port (default port 5060), and the Audio Port range (default ports 9000-9255) pass.
- If the firewall check fails, you must go to your firewall and troubleshoot why the test failed.
Please Note: Neither Imecom nor 3CX provide specific firewall configuration support. 3CX provides a page on their website that focus on configurations for popular firewalls.
Step 3: Add the handSIP Account in 3CX Phone System
Once your handSIP account has been created, you will need to configure the account in 3CX Phone System.
- In the 3CX Management Console menu, select SIP Trunks > Add SIP Trunk.
- Select United States.
- Select handSIP from the Provider drop down list. Important: If handSIP is not listed, select the “Generic” option in the Country drop down menu and then choose between either Generic VoIP Provider or Generic SIP Trunk.
- Enter the DID assigned to the handSIP SIP Trunk. If you have more than one DID, you can select any one of the DIDs as the main number. Once you’ve chosen and entered your main number, click OK. The SIP Trunk will be created and a new dialog will open.
- Enter “handSIP” as the name for the VoIP provider account. The SIP server hostname or IP and optional Outbound Proxy should be filled in automatically. If they are not, please add the following:
Specify the number of simultaneous calls. The number of simultaneous calls is included in the handSIP welcome email containing your demo or paid account details.
In Authentication, if you selected the handSIP VoIP Provider template, this will be automatically filled in and you must leave as is. If not, select Account/Registration. The outbound or inbound only are not applicable and can be ignored.
Specify how calls to the main number should be routed. The routing configured here will be for calls matching the main number.
If you have DID numbers, you will need to specify these in the DIDs tab. Click on the DIDs tab and add the DID number(s) associated with your handSIP account. The DID will be created and linked to the handSIP operator extension. You can change this later from the Inbound Rules node by adding an inbound rule for the DID and routing to the desired destination.
In the Caller ID tab, add the caller ID you wish to have appear on outbound calls.
Click OK to save the trunk settings.
- SIP Server Hostname or IP: sip.handsip.com
- Outbound Proxy: sip.handsip.com
Step 4: Create an Outbound rule to route calls over the handSIP SIP Trunk
Once the handSIP account has been created and configured in 3CX, the next step is to create an outbound call rule.
- Go to the Outbound Rules node and click Add to create a new rule.
- Decide what calls should be routed over this trunk.
- In the Make Outbound Calls section, select the handSIP trunk you just created.
- Click OK to create the outbound rule.
- See 3CX documentation for more information on creating an outbound rule in 3CX.
Please Note: If you are routing calls through a gateway connected to your 3CX PBX, consult the official 3CX documentation for step-by-step instructions on configuring your gateway.