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Installing SIP.US Module in AsteriskNOW

Starting at the FreePBX GUI --

Go to the "Admin" tab and select "Module Administration." Select "Upload Module" and input https://login.sip.us/sipus-2.1.3.tgz in the download field. Then click "Download [From Web]

When the download is complete, you should see a screen similar to:

Now it's time to install the module, as you can see below, this doesn't happen automatically:

Expand the SIP.US section and click on "Install."

Then you should see a screen similar to this. Click on "Confirm."

At this point a status screen should appear. When it's finished, you will need to click on "Return" and click "Apply Config" if it shows up at the top.

Now it's time to go to the "Connectivity" tab and look for "SIP.US." If it's present the module is installed successfully and you should click on this to continue configuring connectivity.

Once you are inside of the module, you will need to select "Add SIP.US Account" and a box will pop up asking for your "Configuration Key." Your key can be found by logging into your SIP.US account, going to "Sip Trunking" and then selecting "FreePBX Config." On that page you will find a link that says "To view your FreePBX configuration key click HERE." When you click on that link you will be shown your key and you can copy it directly into the box shown in the image below. Click "Add" and then wait for the information transfer to complete.

Once the transfer is complete your screen should look similar to the image below. Check both boxes under "Trunks," click "Save," and then after the save completes click "Apply Config":

At this point you should have a "Trunk" on the right hand side of the module (Typically Trunk1). When you select that trunk it should take you to a screen similar to the one below. Note that at the bottom of this screen is where inbound calling is controlled. If you already purchased and selected DID's they will appear here (or if this is a free trial you can borrow a DID for four hours by using the red box on the right side of your SIP.US account screen). If your DID's aren't showing up, you may need to click on "Refresh SIP.US DID List." Once your DID's are imported into the module you need to note that they haven't been directed to go to any destination.

To ensure calls to these DID's are routed properly, simply select the drop down for each DID under "Route To:" and pick an appropriate destination for the calls to be delivered to when the corresponding DID is dialed. Click Save" then "Apply Config."

At this point you have successfully configured inbound and outbound calling. Make some test calls in both directions! If you are experiencing any trouble, be sure to open a support ticket by clicking on the link in the upper righthand corner of your SIP.US account screen.

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