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Asterisk Admin GUI is an open source interface for configuring the Asterisk PBX server. The Asterisk Admin GUI interface can vary slightly depending on which distribution you use. This may be downloaded from the Asterisk Admin GUI website or through one of the major Asterisk distributions such as trixbox, Elastix, PBX in a Flash, etc...

Using the Asterisk Admin GUI 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 via the GUI Interface. Below we provide some resources which you can visit to obtain further information.

Please note, Callcentric is not responsible for preventing unwanted physical or remote access to your IP PBX. If your IP PBX is compromised, you will be responsible for all associated damages.

Please be sure to read this guide regarding securing your IP PBX solution.

RESOURCES

Websites
trixbox - http://www.trixbox.org
Elastix - http://www.elastix.org
pbx-in-a-flash - http://pbxinaflash.net

Help / Support
PBX in a Flash support

Configuring an Asterisk PBX using the Asterisk Admin GUI interface

Here we will configure Asterisk through the Asterisk Admin GUI administrative interface to properly route both incoming and outgoing calls to and from Callcentric. This guide assumes that you have installed Asterisk Admin GUI using either the Asterisk Admin GUI package (or distro), trixbox, Elastix, PBX in a Flash or a method of your choice. This guide also assumes that the Asterisk Admin GUI install steps were completed properly and that you have administrative access to the Asterisk Admin GUI administration interface.

This guide is based on version 2.11 of Asterisk Admin GUI.

We recommend that you read each step thoroughly and in its entirety before performing the action(s) 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/updates 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 11.5.1 built by root @ server on a i686 running Linux on 2013-09-19 14:01:43 UTC

STEP 1 Trunk Configuration

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 Asterisk Admin GUI administrative interface
From the navigation bar, click on Connectivity >> Trunks
Click Add SIP Trunk in middle of page
In the General Settings section; locate the Trunk Name option, and specify callcentric on the given field
In the Outbound Settings section; Locate the Trunk Name option, and specify callcentric on the given field

Copy and paste the following into the PEER Details field.

For asterisk 1.2 based PBX, please use the following:

context=from-pstn-toheader
fromdomain=sip.callcentric.net
fromuser=1777MYCCID
host=sip.callcentric.net
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-toheader
fromdomain=sip.callcentric.net
fromuser=1777MYCCID
host=sip.callcentric.net
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-toheader
fromdomain=sip.callcentric.net
fromuser=1777MYCCID
host=sip.callcentric.net
insecure=port,invite
secret=SUPERSECRET
type=peer
defaultuser=1777MYCCID
canreinvite=no
videosupport=no
disallow=all
allow=ulaw

For asterisk 1.6.2 / 1.8 / 10 and 11 based PBX, please use the following:

context=from-pstn-toheader
fromdomain=sip.callcentric.net
fromuser=1777MYCCID
host=sip.callcentric.net
insecure=port,invite
secret=SUPERSECRET
type=peer
defaultuser=1777MYCCID
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw

Scroll down to the Registration section
Enter your Register string in this format:
```
1777MYCCID:[email protected]
```
*** Where 1777XXXXXXX is either the default extension 1777MYCCID OR 1777MYCCIDEXT, where 1777MYCCID is the 1777 number assigned to you by Callcentric and EXT is the three digit extension you are trying to register this UA to.

*** Where SUPERSECRET is your extension SIP Password here. Your extension SIP password is the password you created for the extension you are trying to use. You may edit the SIP password you wish to use in by logging into your My Callcentric account and clicking on the Extension menu link and then modifying the appropriate extension.
Click on Submit Changes to add your new SIP trunk to your Asterisk server
Click on the red button labeled Apply Config at the top of the screen to apply the changes you just made

Now you will want to edit your sip_general_custom.conf file and enter, or modify, the following lines:
```
context=from-pstn-toheader
srvlookup=yes
session-timers=refuse
```
If using trixbox/Elastix this will have to be done through the web interface to edit your config files.
If using PBX in a Flash, you will need to log in to your server and edit the /etc/asterisk/sip_general_custom.cond file manually, usually with an editor such as nano.
If you've installed the Asterisk SIP Settings module (from the navigation menu, click on Settings >> Asterisk SIP Settings), please set the following options within the Advanced General Settings section:

Other SIP Settings

Default Context from-pstn-toheader

SRV Lookup Enabled

session-timers refuse

STEP 2 Outbound Route Configuration

An outbound route sends calls which are dialed in a certain pattern to your desired provider, in this case Callcentric.

From the navigation bar, click on Connectivity >> Outbound Routes, to configure your Asterisk box to send calls to Callcentric
Enter to-callcentric into Route Name field
Locate the Dial Patterns that will use this Route section, and specify the following options:

Prepend

Prefix 9

Match Pattern .

CallerID
Locate the Trunk Sequence for Matched Routes section, and select the callcentric trunk from the drop down list
Click on Submit Changes to add your new route to your Asterisk server
Click on the red button labeled Apply Config at the top of the screen to apply the changes you just made

STEP 3 Extension Configuration

We will need to create a local extension on your Asterisk PBX. A local Asterisk extension 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.

From the navigation bar, and click on Applications >> Extensions to add a new extension which will connect to your Asterisk server
From the drop-down menu, select Generic SIP device

Enter 1000 as the User Extension
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 button labeled Apply Config at the top of the screen to apply the changes you just made

STEP 4 Inbound Route Configuration

Inbound Route 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, such as DID-Bbased routing.

If you have already configured an extension then you may substitute your pre-configured extension for point 4 below.

From the navigation bar, click on Connectivity >> 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 in order to match any incoming call. This is useful if you wish to receive all calls
Scroll down to the Set Destination section
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 button labeled Apply Config at the top of the screen to apply the changes you just made

STEP 5 Configure and test UA

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).

STEP 6 Placing Test Calls

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.