Getting started with FreePBX – Part 3 Making external calls

This article assumes you have completed part 1 and part 2

So we have configured our trunk and our extension and now we need to tell FreePBX to use that trunk when someone dials a number. This is done by creating an Outbound Route.

The configuration below also takes in to consideration the fact that I am in the UK and that callcentric is a US company. Your setup may differ.

Creating an outbound route

First we need to click on “Outbound Routes” in the main menu on the left hand side of the screen

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Now we need to create the route. Give it a name – I’ve called mine “external”

Now we need to enter a “Dial Pattern”. This is what decides if this route is used or not.  You can hover your mouse over “Dial Patterns” to get a description.

I have entered “0.” as my dial pattern. This means that if an extension/user dials any number beginning with 0 (zero) it will use the trunk selected.

Now select “SIP/callwithus” to be your destination trunk and click “Submit Changes”

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Now, because I’m in the UK I’m going to create a “Dial Rule” on my trunk. So click on “Trunks” on the left hand menu and select our trunk “SIP/callwithus” on the right hand menu.

I’ve created 1 rule and this is what it does -

0044+0|Z.  – if someone dials a number that starts with 0 followed by another number that’s not a 0 then remove the first zero and adds 0044 to the front of the number. Basically this turns a local UK number into the international equivalent. So “0800 123 4567″ becomes “0044 800 123 4567″.  I need this because callwithus is a US based provider and needs to be told to dial a UK number as an “International” number.

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Scroll down to the bottom and click “Submit Changes”

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Now you should be able to go to your extension and dial an external number. Either a national UK one or an international  number, by using 00 as the international prefix

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Related posts:
  1. Getting started with FreePBX – Part 6 Cheap phone calls using DISA and Callback
  2. Getting started with FreePBX – Part 1 Setting up a trunk
  3. Getting started with FreePBX – Part 4 Setting up a DID number



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10 Comments

  1. sana:

    Hi,

    I have a quick question. According the configuration my asterisk server running on 192.168.1.102. I can successfully communicate internal calls via X-lite. However, I cannot make any outbound call or inbound call from ipkall. Do I need to do anything about my home router to call make outside? For ipcall configuration, I gave my IP address to SIP proxy but cannot revived call. Server is connected through lynksis WRTSL54GS router. Thank for your help.

  2. matt:

    There are a couple of things you probably need to do. The first is to tell the outside world what your external IP is. You can do that with the externip command, see here for details – http://www.voip-info.org/wiki/view/Asterisk+SIP+externip The second is to forward some ports in your firewall. You will probably need ports for SIP and RTP forwarded to your Asterisk server. You should be able to figure out the ports you need here – http://www.voip-info.org/wiki/view/Asterisk+firewall+rules.

    Running an Asterisk server behind a NAT firewall can definitely be a pain!

  3. sana:

    Hi Matt:

    Thanks for your help. I have successfully configured NAT and firewall, then, I can call from IPKALL number(inbound). However, I cannot call from my x-lite to outbound. I have did same config. as u did for callwithus. Gave me – Call fail: 503 service unavailable. Do you have any idea? . My balance is 0 in callwitus account. Thanks for your all support.

  4. matt:

    Did you try adding some funds to you callwithus account. I’m not sure what numbers you can call with a zero balance?

  5. jordan diment:

    Dear All.

    I am using free PBX and want to block mobile calls so only allow 01, 02 to be made on the system.
    can anyone please advise the best way to do this?

    Cheers

    Jordan

  6. matt:

    Hi Jordan,

    You can easily do this on your Outbound Routes in FreePBX. Check out the help for ‘Dial Patterns’ as these are what decides which calls are completed using that route.

  7. Haitham:

    Hi,

    I want to ask is there any way to use ordinary modem 56K to interface analog lines in freepbx?

  8. matt:

    Not to run the voice part of the call over no. You would need either an FXO card for your server, or an FXO gateway.

    If you mean to run the voip part of the call over (instead of broadband) then you could but the latency would probably be too high.

  9. Haitham:

    Thank you for your reply
    What I need is to make calls from sip phones “X-lite” to analog phones and receive calls from analog phones to either sip phones or analog phones.
    I checked many pages but I got confused. Is the ordinary computer modem is capable to perform this function, even with weak performance? I just want to test this to take decision whether I will use the freepbx or not.
    Thanks a lot

  10. matt:

    No, I’m afraid that will not work. You need an fxo card/gateway.

    Or you could look at a SIP service such as voip.ms. FreePBX connects to an ITSP like this and will pass and receive calls to the PSTN, normal telephone network.

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