A Guide to Connecting to FreePBX FreePBX is a basic web Graphical User Interface that manages Asterisk PBX. It includes many features available in other PBX systems such as voice mail, conference calling, IVR (phone menus), and automatic call distribution. For the purpose of this example. The IP of our FreePBX installation is: 123.45.67.890 (in the real world, this IP could never exist due to the 890 being higher than the value.255) In this tutorial we will cover how to make outbound and inbound calls to/from Aloha Connect.
Outbound Calling The first thing you will need to do to make calls to the outside world is to set up a SIP trunk. Log into aloha- connect and go to Connecting > VOIP access. Clicking VOIP Access will take you to the screen shown above. SIP URI: Is the destination where you will send the call attempt. SIP Username: Your Aloha Connect Account Number SIP Password: Your Password as defined on the page. Press 'Reset SIP Password' to be randomly generated a new password SIP Call Limit: The amount of concurrent calls you can make. Please contact us to increase this for you. You will need to take a note of these details. For the next part, open the FreePBX admin panel, on the sidebar click on trunks, then add SIP Trunks. Give your trunk a name (any descriptive name). We recommend you leave the rest of the General Settings and the section Dialled Number Manipulation Rules as default until you are more familiar with this system. On the section, Outgoing Settings you will need to edit Peer details in the manner shown on the left. Delete everything in the user Details block. For register string, the protocol is username:password@host. In this case that means 45861:5161552159@sip.aloha- connect.com
When you are happy press submit changes, then apply configuration changes (should be highlighted orange) To test that everything has been setup up correctly, you can goto the home page by pressing admin at the top of the screen and looking that IP Trunk Registrations is green meaning that your system has successfully been able to register to the Aloha Connect Platform. You will now need to set up an outbound route, to do this click on outbound routes, then add route. Name the route and place a full stop in the match pattern field. For trunk sequence place the trunk you have just created in the dropdown box. The '.' in the dial pattern is telling the system that anything dialled will be routed through the defined trunk which we have called 'aloha'. Connecting an IP phone To connect a phone to the system you must first create an extension on FreePBX. For the purposes of this manual a connection to a free version of Xlite 4 will be described, though principles for other software is similar, to do this click on extensions and create a generic SIP device. The user extension is the unique number of your phone; this field must be numbers only. The display name is unimportant, but we recommend it be descriptive to make use easier. Outbound caller ID is the number shown to users. This will show on internal calls. If you have paid for the feature with Aloha where we allow you to pass your own CLI out, then this is the section that you edit to show a CLI (Caller ID) for that extension. A secret (password) is required, we highly recommend that this password contains a mixture of upper/lower case characters.
Leave other fields blank, some features may be altered without affecting connections, but are not necessary for basic use. When you are satisfied with this page, click submit, then apply configurations The next part of the process is to load up Xlite. Go into preferences and the panel on the left should appear. Enter your user ID (which will be your extension), the IP address of the server running FreePBX the password (secret). Everything else can be left as default for now. You should now be able to make outbound calls through aloha- connect. Testing Call our free test numbers below. This is a great way to check a) the latency of the call (i.e how long it takes for the call to go from your phone to our system back to yours) and b) DTMF (the codes you enter on the phone i.e press 1 for sales, 2 for support etc) Latency (ECHO Test) Number: 03303 501 251 DTMF (Number Pressing Test): 03303 501 250
Summary for making outbound calls with Aloha Connect in FreePBX. - Setup the Aloha Trunk (make sure you fill in the register string as well. Not just the Peer Details) (Do NOT touch the CLI section here) - Setup the Outbound Route to use the Aloha Connect Trunk (Do NOT touch the CLI section here) - Setup an extension where you have defined the CLI for this extension that will be shown to the caller if you have signed up for the Customer CLI Service. - Hook up to a softphone such as X- Lite or Zopier to make a test call. NOTE: Due to anti fraud measure, Aloha will send signalling to answer all calls. You will only be billed for actual answered calls. So CDR's (Caller record) in FreePBX will show time Inbound Calls This section describes how to setup Global Numbers on Aloha Connect to be redirected to your FreePBX Installation via SIP. In FreePBX you will need to goto Tools > Asterisk SIP Settings Scroll to the bottom of the page till you see 'Allow SIP Guests' and change this to yes. This option will allow unauthenticated inbound calls to FreePBX. Click on Submit Changes and then reload asterisk. After you have done this. Login to Aloha connect and goto Global Numbers (Connecting > Global Numbers) Above is a list of numbers we have in our demo account. Today, we are going to take the number 442922190325 which is current setup to redirect calls to 08000480202 and redirect direct it to our FreePBX Server. Remember the IP of our demo server is 123.45.67.890.
To update the number, simple click on the pencil to load up the edit page. To redirect the number to your PBX via SIP. Simply press 'Phone Number' and write the following in this format SIP: 442922190325@123.45.67.890 Where SIP is the protocol to be used. 442922190325 is your DID in international format and 123.45.67.890 is the IP of your FreePBX installation. Aloha connect supports domain address (i.e yourname.com) instead of IPs. The next step is to setup this number up in FreePBX. Do this by clicking on 'Inbound Routes' in the left side menu. You then simply enter the number in international format in the DID Number box and put a description for the number in the description box. For basic use the other boxes do not need to be touched. At the bottom of the page you will see a box called set destination Here you will set the destination of where the call is to be routed. In this case, I want calls to goto the extension I setup earlier.
Simply select Extensions > and the extension I created earlier. Press submit, reload asterisk and now all calls to that number will now be redirected to that extension. Inbound Summary - Update the Global Numbers number to be redirected in the SIP:DID@IP format (where DID is your number in international format (without the plus) - Allow SIP Guests in your SIP settings for FreePBX - Add the inbound number to your PBX and select the destination for the number to be routed to.