Supported devices
ATA Adaptors
IP Phones
Cordless IP Phones
IP Conference Phones
IP PBX Software/Hardware
Desktop Softphones
Mobile Softphones

Security considerations for Callcentric customers

3CX Version 6
3CX Version 12
3CX Version 12.5
3CX Version 15
3CX Version 16
3CXPhone (for Windows)
3CXPhone (Mobile)
Aastra 6753i
Acrobits/Groundwire for iPhone
Android SIP Client
Apivio MWP1100
Asterisk 1.4 and 1.2
Asterisk 1.6
Asterisk 1.6.2, 1.8, and 10
Asterisk 14
Asterisk 17 CHAN_SIP (Vanilla)
Asterisk 17 PJSIP (Vanilla)
Asterisk Admin GUI v2.11
Asterisk Admin GUI v12
Asterisk Admin GUI v13
Asterisk Admin GUI v15
Bria Solo
Bria Desktop
Bria Mobile
Cisco ATA 186/188
Cisco SPA112/SPA122
CloudTC Glass 1000
D-Link DVG-1402S
Gigaset A510 IP
Gigaset C610A IP
Gigaset DX800A
Grandstream DP715/710
Grandstream DP750/720
Grandstream DP752/730/722
Grandstream GAC2500
Grandstream GRP2612
Grandstream GRP2613
Grandstream GRP2614
Grandstream GRP2615
Grandstream GXP1450
Grandstream GXP1620
Grandstream GXP1625
Grandstream GXP1630
Grandstream GXP1760
Grandstream GXP1782
Grandstream GXP2110
Grandstream GXP2130
Grandstream GXP2140
Grandstream GXP2160
Grandstream GXP2170
Grandstream GXP2200
Grandstream GXV3140
Grandstream GXV3240
Grandstream GXV3275
Grandstream GXV3370
Grandstream GXV3380
Grandstream HandyTone 286
Grandstream HandyTone 486
Grandstream HandyTone 702
Grandstream HandyTone HT802
Grandstream HandyTone HT814
Grandstream HandyTone HT818
Grandstream UCM6102
Grandstream UCM6204
Grandstream Wave Lite
Grandstream WP820
Htek UC803
Htek UC860
Htek UC924
Htek UC926
Innomedia SIP MTA-6328
Innomedia BuddyTalk 110
Linksys PAP2
Linksys SPA3102
Linphone Desktop (v4.1.1)
Linphone Desktop (v3.4.3)
Linphone Mobile
Linphone Mobile (v.3.2.3)
NCH Express Talk
Nokia E90
pbxnsip Version 3
Polycom SoundPoint IP 601
Polycom SoundStation IP 5000
snom 1xx/2xx/3xx
snom 820
snom D717
snom D735
snom D785
snom M9
snom ONE
Telco AC-211
trixbox / Elastix / pbx-in-a-flash
Windows Messenger
Unidata ICW1000G
Uniden EXP1240
UTStarcom F3000
Vodia PBX (v5)
Vodia PBX (v64.0)
VTech VCS754
VTech VDP650
VTech VSP600
VTech VSP725
VTech VSP726
VTech VSP735
VTech VSP736
X-Lite / X-Pro / eyeBeam
Yealink T32G
Yealink T41S
Yealink T42G
Yealink T42S
Yealink T46G
Yealink T46S
Yealink T48S
Yealink W52P
Yeastar MyPBX U100
ZoIPer 3.2
ZoIPer 5
ZoIPer Mobile
Zoom 5801
Zycoo CooVox U20
Other Linksys/Sipura products
Generic / Other Device

DID-Based Routing with Asterisk
Asterisk Admin GUI v13

Setup information for other versions:
Asterisk Admin Gui version 15
Asterisk Admin Gui version 12
Asterisk Admin Gui version 2.11

Tree FrogAsterisk 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 directly from the Asterisk Admin GUI website or through one of the major Asterisk distributions such as trixbox, Elastix, PBX in a Flash, etc...

With 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 in a more manageable interface. Below we provide some resources 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.


Help / Support
Asterisk entry via Voip-info
Elastix support
PBX-in-a-Flash support

Configuring Asterisk PBX (chan_sip) 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), Elastix, IncrediblePBX 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 13.0.113 of the Asterisk Admin GUI (running Asterisk 13.7.1)

We recommend that you read each step through 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 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", and you should see an output similar to the following:

Asterisk 13.5.0 built by root @ server on a i686 running Linux on 2015-08-10 13:54:19 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. To configure a SIP Trunk, please proceed with the following:

  1. Login to Asterisk Admin GUI administrative interface

  2. From the navigation bar at the top of the page, click on Connectivity >> Trunks

  3. Click the Add Trunk button that is located in the middle of page, and select Add SIP (chan_sip) Trunk from the drop down menu

  4. In the General section, locate the Trunk Name option, and specify callcentric in the given field

  5. Click on the SIP Settings tab, and click on the Outgoing sub-section tab

  6. Locate the Trunk Name option, and specify callcentric in the given field

  7. Copy and paste the following into the PEER Details field.


  8. Click on the Incoming sub-section tab

  9. 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.

  10. Click on Submit Changes to add your new SIP trunk to your Asterisk server

  11. Click on the Apply Config button at the top of the screen to apply the changes you've just made

  12. From the navigation menu, click on Settings >> Asterisk SIP Settings

  13. From the sub section General SIP Settings, locate the option Allow Anonymous Inbound SIP Calls, and set this option to No

  14. Click on Submit Changes to save your changes

  15. Click on the Apply Config button at the top of the screen to apply the changes you've just made

  16. From the sub-menu bar, click on Chan SIP Settings tab

  17. Locate the option Allow SIP Guests, and set this option to No

  18. Locate the option SRV Lookup, and set this option to Yes

  19. Locate the option Other SIP Settings, and use the following settings:

    sendrpid = yes
    trustrpid = no
    disallowed_methods = UPDATE
    session-timers = refuse

  20. Click on Submit Changes to save your changes

  21. Click on the Apply Config button at the top of the screen to apply the changes you've just made

  22. From the navigation bar at the top, visit the Admin >> Config Editor (this assumes that you've installed the Config Editor module on your Asterisk Admin GUI installation. If you have not installed it, you can install it by visiting the Admin >> Module Admin configuration page)

  23. On the left pane, locate the file labeled sip_custom_post.conf and copy and paste the following into that file:


    ***** The value that you've specified in the option Outgoing settings >> Trunk Name within your trunk configuration page, must be specified within the parentheses on the settings above; so if you've used "callcentric" (all lowercase) for your Trunk Name (as presented in our example), please specify the above settings as-is. Or for example if you've specified Trunk_1 for that option, you will need to specify the following:


    Your configurations should look similar to the screenshot below:

  24. Click on Save to save the file above

  25. Click on the red button labeled Apply Config at the top of the screen to apply the changes you just made

* If you do not have the Config Editor module installed, you will need to log in to your server and edit the /etc/asterisk/sip_custom_post.conf file manually, usually with an editor such as nano.

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.

  1. From the navigation bar, click on Connectivity >> Outbound Routes, to configure your PBX to route outgoing calls towards your Callcentric trunk

  2. From the sub-section Route settings, enter to-callcentric into the Route Name field

  3. Locate the Trunk Sequence for Matched Routes section, and select the callcentric trunk from the drop down list

  4. From the sub-menu bar, click on Dial Patterns

  5. Locate the Dial Patterns that will use this Route section, and specify the following options:

    match pattern.

  6. Click on Submit Changes to add your new route to your Asterisk server

  7. Click on the Apply Config button at the top of the screen to apply the changes you've just made

STEP 3 Extension Configuration
In this step, we'll create a local extension on your PBX. This local extension (on your PBX) provides an account number that another User Agent (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.

  1. From the navigation bar, and click on Applications >> Extensions to add a new extension which will connect to your Asterisk server

  2. From the drop-down menu, select Generic SIP device (or Add New Chan_SIP extension)

  3. Enter 1000 as the User Extension

  4. 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

  5. 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

  6. Click on Submit Changes to add your new extension to your Asterisk server

  7. Click on the Apply Config button at the top of the screen to apply the changes you've just made

STEP 4 Inbound Route Configuration
With an inbound route you are given the flexibility to send incoming calls to a wide 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 call on ANY numbers, 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 Based Routing.

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

  1. From the navigation bar, click on Connectivity >> Inbound Routes to configure the routing of calls to your Callcentric account.

  2. If there isn't a default inbound route defined already within your PBX, then click on Add Incoming Route. You will first want to fill the DID Number field with your 1777 number, or if you've acquired a phone number from us already, please use your phone number in the DID Number field (i.e. if you've acquired the number 12125551000, please use 12125551000 on this field). Make sure to leave the Caller ID Number blank in order to match any incoming call. This is useful if you wish to receive all calls

  3. Scroll down to the Set Destination section

  4. Choose First Extension (1000) from the Core drop down box

  5. Click on Submit Changes to add your new inbound route to your Asterisk server

  6. Click on the Apply Config button at the top of the screen to apply the changes you've just made

STEP 5 Configure and test UA
  1. Choose your desired UA

  2. 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

  3. 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
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.