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Supported devices |
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| freePBX / trixbox |
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| GENERAL INFORMATION |
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freePBX is an open source interface for configuring the Asterisk PBX server. The freePBX interface can vary slightly depending on which distribution you used. This may be directly from the freePBX website or through the many of the major Asterisk distributions such as trixbox, Elastix, or PBX in a flash, etc...
Using freePBX you are able to configure most of Asterisk's options without editing the individual configuration files. You can also setup advanced options such as call routing, voicemail and other calling features in a more manageable interface. Below we provide some resources which you can visit to obtain further information.
DID based routing with trixbox / freePBX
Please note that Callcentric is not responsible for preventing unwanted physical or remote access your IP PBX. If your IP PBX is compromised then you will be responsible for any damage caused.
Please be sure to read this guide regarding securing your IP PBX solution.
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| RESOURCES |
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Main Project Pages:
freePBX - http://www.freepbx.org
trixbox - http://www.trixbox.org
Help / Support:
freePBX support page
Aussie VoIP freePBX documentation
trixbox Forum
trixbox Documentation
Setup Guides:
Nerd Vittles trixbox and freepbx 2.1.1 Guide
VoIP-info.org trixbox Wiki |
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| Configuring the Asterisk PBX using the freePBX interface |
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Here we will configure Asterisk through the freePBX administrative interface to properly route both incoming and outgoing calls to and from Callcentric. This guide assumes that you have installed freePBX using either the freePBX package, trixbox or a method of your choice. This guide also assumes that the freePBX install steps were completed properly and that you have administrative access to the freePBX administration interface.
We recommend that you read each step through in its entirety before performing the action indicated in the step.
We also recommend that you check which version of Asterisk your PBX is based on; as there are many significant changes between each revision of Asterisk. To check which version your PBX is based on; please log into your PBX's command line interface, and execute the command "show version" or "core show version" you should see an output similar to the following:
Asterisk 1.8.7.1 built by root @ freepbx2_9 on a i686 running Linux on 2011-11-26 16:06:53 UTC |
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| STEP 1 |
Trunk Configuration |
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In the context of this guide a trunk is used to route calls between your Asterisk PBX and your desired VSP(Voice Service Provider), in this case Callcentric. In this section we will configure a SIP trunk.
- Login to freePBX administrative interface
- Click on Setup in top right of page
- Click on Trunks in left side navigation
- Click Add SIP Trunk in middle of page
- Scroll to Outgoing Settings and enter callcentric into
Trunk Name field
- Copy and paste the following into the PEER Details field.
For asterisk 1.2 based PBX, please use the following:
context=from-pstn
fromdomain=callcentric.com
fromuser=1777MYCCID
host=callcentric.com
insecure=very
secret=SUPERSECRET
type=peer
username=1777MYCCID
disallow=all
allow=ulaw
For asterisk 1.4 based PBX, please use the following:
context=from-pstn
fromdomain=callcentric.com
fromuser=1777MYCCID
host=callcentric.com
insecure=port,invite
secret=SUPERSECRET
type=peer
username=1777MYCCID
canreinvite=no
videosupport=no
disallow=all
allow=ulaw
For asterisk 1.6 based PBX, please use the following:
context=from-pstn
fromdomain=callcentric.com
fromuser=1777MYCCID
host=callcentric.com
insecure=port,invite
secret=SUPERSECRET
type=peer
defaultuser=1777MYCCID
canreinvite=no
videosupport=no
disallow=all
allow=ulaw
For asterisk 1.6.2 / 1.8+ based PBX, please use the following:
context=from-pstn
fromdomain=callcentric.com
fromuser=1777MYCCID
host=callcentric.com
insecure=port,invite
secret=SUPERSECRET
type=peer
defaultuser=1777MYCCID
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw
Where 1777XXXXXXX is your Callcentric number and SUPERSECRET is the same password you create when you sign up for you Callcentric account. Optionally you may change it in your preferences.
- Scroll down to Registration
- Enter your registration string in this format:
1777MYCCID:SUPERSECRET@callcentric.com/1777MYCCID
Where 1777XXXXXXX is your Callcentric number and SUPERSECRET is the same password you create when you sign up for you Callcentric account. Optionally you may change it in your preferences.
- Click on Submit Changes to add your new SIP trunk to your Asterisk server
- Click on the red bar at the top of the screen to apply the changes you just made
View screenshot
- Now you will want to edit your sip_general_custom.conf file and enter, or modify, the following lines:
context=from-pstn
srvlookup=yes
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas
If using trixbox this will have to be done through the web interface to edit your config files.
View screenshot
If using freePBX you will need to log in to your server and edit the /etc/asterisk/sip.conf file manually, usually with an editor such as nano.
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| STEP 2 |
Outbound Route Configuration |
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An outbound route sends calls which are dialed in a certain pattern to your desired VSP, in this case Callcentric.
- Click on Outbound Routes to configure your Asterisk box to send calls to Callcentric
- Enter to-callcentric into Route Name field
- Scroll to Trunk Sequence and select the SIP/callcentric trunk from the drop down list
- Click on Submit Changes to add your new route to your Asterisk server
- Click on the red bar at the top of the screen to apply the changes you just made
View screenshot
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| STEP 3 |
Extension Configuration |
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An extension in this context is an account on your Asterisk PBX which provides an account number which another UA (software or
hardware used for calling) can connect to in order to make and receive calls. There are a few types of extensions. Here we will
create a SIP extension.
If you have already configured an extension then you may skip this step. Then in the next step (Inbound Route Configuration) you may
use your pre-configured extension.
- Click on Extensions to add a new extension which will connect to your Asterisk server
- Choose SIP as the extension type
- Enter 1000 for the extension number
- For now we will use a generic identifier for this extension. Enter First Extension
for the Display Name field. Later you may enter a unique identifier of your choice
- Enter your desired password in the Secret field. You will use this password when configuring your
desired UA later in order to connect to your Asterisk PBX
- Click on Submit Changes to add your new extension to your Asterisk server
- Click on the red bar at the top of the screen to apply the changes you just made
View screenshot
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| STEP 4 |
Inbound Route Configuration |
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Inbound configuration can become extremely complex. With an inbound route you are given the flexibility to send incoming calls to a whole
range of destinations. For example you may route an incoming call to a specific extension, to a ring group or to an IVR. In this section
we are going to setup an inbound route which will handle ANY incoming calls on ANY number, including emergency numbers, and simply route
those calls to a specific extension (1000). Later on you can configure more complex routing schemes.
If you have already configured an extension then you may substitute your pre-configured extension for point 4 below.
- Click on Inbound Routes to configure the routing of calls to your Callcentric account
- If there isn't a default inbound route called 1777MYCCID / any CID then click on
Add Incoming Route. You will first want to fill the DID Number
field with your 1777 number. Make sure to leave the Caller ID Number
and Zaptel channel blank in order to match any incoming call. This is useful if you wish to
receive all calls
- Scroll down to Set Destination
- Choose First Extension (1000) from the Core dropdown box
- Click on Submit Changes to add your new inbound route to your Asterisk server
- Click on the red bar at the top of the screen to apply the changes you just made
View screenshot
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| STEP 5 |
Configure and test UA |
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- Choose your desired UA.
- Use the IP address or hostname for your Asterisk box along with 1000 (the extension created earlier) and password for the 1000
extension to connect to your Asterisk box
- Next you will want to try placing test calls to and from your Asterisk PBX using the UA currently connected to your newly
created extension (1000).
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| STEP 6 |
Placing Test Calls |
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You can make a test call to 17771234567, or if you are signed up for one of Callcentric's rate plans you
can place a call to a traditional landline or mobile phone by dialing either:
1 + the area code and number for calls to the US
Or
011 + the country code, area code, and number for calls worldwide
(you may also use 00 instead of 011).
To test inbound calls from Callcentric to your Asterisk installation, follow the directions
listed in this FAQ. |
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