In this series of articles I will run through how to get started once you get FreePBX setup. You will need to run through the articles in order as some of the later ones will rely on items set up in earlier articles.
For a trunk (required to make calls to the outside world) I will use callwithus. Click here for a free account.
Adding a trunk
The main FreePBX menu is down the left hand side of the screen
Click Trunks
This will bring up the “Add a Trunk” page
Now there is a sub-menu on the right and side of the page. Click “Add Trunk” there.
Now click “Add SIP trunk”
Give the trunk a name in the “Trunk Name” box
Enter your callwithus details in the “PEER Details” box as shown below. Obviously using your account number and secret (password)
Enter the register string in the “Register String” box as show below, again using your username:password (they are separated by a colon)
And the click “Submit Changes”
*** PLEASE NOTE – YOU SHOULD NOW BE USING ‘sip.callwithus.com’ RATHER THAN ‘uk.callwithus.com’ ***
Whenever you submit any changes in FreePBX you then have to click on the “Apply Configuration Changes” before the settings will take effect
We can now check that everything looks good with our trunk
Click on the “Tools” tab at the top of the main menu
Click on “Asterisk Info” in the main menu
Now click on “Sip Info” on the right hand menu and we should be able to see our trunk registered
And that should be our trunk setup and ready to use.
In the next article we will setup an extension so we can try making some calls via our trunk
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Hi: I am using CWU but I have a problem with sound transmission. When I call a cellphone, the other party does not hear me, even though I hear them. I am not sure what could be the problem. I am using hosted asterisk server.
I have seen this before. Try ensuring you have all codecs enabled for the CWU trunk. Also try forcing different codecs.
If that doesn’t work I would raise a call with them.
Hi,
I have configure the xlite and arstrisk server on my pc and i am ableto connect x-lite form my
Asterisk server.
But when i try to call on my number it show “Not acceptahle here”
any other number also give the same response.
can any one tell me on which number i can call.
With thnaks
Jeet
You could try calling *43. That should be an echo test and will confirm if your extension is set up correctly.
thank you very much. your tutorial is highly explanatory. My question is this, after I followed this tutorial intoto, I tried to connect and dial my GSM phone on from my sip phone, it does not ring but keep showing “Proxy Authentication required.” Is it because I have not bought credit from callwithus that makes me to get this error message? Thanks.
It’s tough to say as there’s no standard SIP code for ‘out of credit’ so providers send back different codes.
However Proxy Authentication Required should mean that callwithus is expecting your device to send them a username/password for SIP authentication and isn’t.