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

Security considerations for Callcentric customers

3CX
3CX Version 6
3CX Version 12
3CX Version 12.5
3CX Version 15
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 Admin GUI v2.11
Asterisk Admin GUI v12
Asterisk Admin GUI v13
Bria Desktop
Bria Mobile
Callcentric Android App
Callcentric iPhone App
Callcentric Softphone
Cisco ATA 186/188
Cisco SPA112/SPA122
CloudTC Glass 1000
CSipSimple
D-Link DVG-1402S
Ekiga
Elastix
Gigaset A510 IP
Gigaset C610A IP
Gigaset DX800A
Grandstream DP715/710
Grandstream DP750/720
Grandstream GXP1450
Grandstream GXP1620
Grandstream GXP1625
Grandstream GXP1760
Grandstream GXP1782
Grandstream GXP2110
Grandstream GXP2130
Grandstream GXP2140
Grandstream GXP2160
Grandstream GXP2200
Grandstream GXV3140
Grandstream GXV3240
Grandstream GXV3275
Grandstream HandyTone 286
Grandstream HandyTone 486
Grandstream HandyTone 702
Grandstream Wave
Htek UC803
Htek UC860
Htek UC924
Htek UC926
Innomedia SIP MTA-6328
Jitsi
Linksys PAP2
Linksys SPA3102
Linphone Desktop
Linphone Mobile
Linphone Mobile (v.3.2.3)
NCH Axon PBX
NCH Express Talk
Nokia E90
Obihai
OBi100/110
OBi200/202
OBi1032
pbx-in-a-flash
pbxnsip
pbxnsip Version 3
PhonerLite
Polycom SoundPoint IP 601
Polycom SoundStation IP 5000
snom 1xx/2xx/3xx
snom 820
snom M9
snom ONE
SFLphone
SJphone
Telco AC-211
trixbox / Elastix / pbx-in-a-flash
Twinkle
Windows Messenger
Unidata ICW1000G
Uniden EXP1240
UTStarcom F3000
Vodia PBX
VTech VCS754
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
ZoIPer 3.2
ZoIPer Mobile
Zoom 5801
Other Linksys/Sipura products
Generic / Other Device

DID-Based Routing with Asterisk
DID-Based Routing with trixbox / Asterisk Admin GUI / Elastix / PBX-in-a-Flash
trixbox / Elastix / pbx-in-a-flash

GENERAL INFORMATION
Trixbox, Elastix, and PBX in a Flash are open source interfaces for configuring an Asterisk PBX server. The Asterisk Admin GUI interface can vary slightly depending on which distribution you use. You can download the Trixbox, Elastix, and PBX in a Flash software directly from their respective websites; to which we have included links below. Using the Asterisk Admin GUI Interface 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.

DID based routing with trixbox / Elastix / PBX in a Flash


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.


RESOURCES
Main Project Pages:
trixbox - http://www.trixbox.org
Elastix - http://www.elastix.org
PBX in a Flash - http://pbxinaflash.net

Help / Support:
trixbox Forum
trixbox Documentation
Elastix support
PBX in a Flash support

Setup Guides:
Nerd Vittles trixbox and Asterisk Admin GUI 2.1.1 Guide
VoIP-info.org trixbox Wiki


Configuring an Asterisk PBX using the Asterisk Admin GUI interface
Here we will configure Asterisk through the Asterisk Admin GUI interface to properly route both incoming and outgoing calls to and from Callcentric. This guide assumes that you have installed the Asterisk Admin GUI using either the Asterisk Admin Gui Package, 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.

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 respective 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 @ ____2_9 on a i686 running Linux on 2011-11-26 16:06:53 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

  • 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 you are using the Asterisk Admin GUI 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.

STEP 2 Outbound Route Configuration
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

STEP 3 Extension Configuration
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

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-Based Routing.

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

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.