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
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
CSipSimple
D-Link DVG-1402S
Ekiga
Elastix
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
Jitsi
Linksys PAP2
Linksys SPA3102
Linphone Desktop (v4.1.1)
Linphone Desktop (v3.4.3)
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 D717
snom D735
snom D785
snom M9
snom ONE
SFLphone
SJphone
Telco AC-211
trixbox / Elastix / pbx-in-a-flash
Twinkle
Windows Messenger
Unidata ICW1000G
Uniden EXP1240
UTStarcom F3000
VitalPBX
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
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 1.4 and 1.2
For Asterisk 17 CHAN_SIP (Vanilla) click here
For Asterisk 17 PJSIP (Vanilla) click here
For Asterisk version 14 click here
For Asterisk version >= 1.6.2, 1.8, 10 click here
For Asterisk version 1.6 - 1.6.1 click here


GENERAL INFORMATION
Asterisk (and Asterisk@Home) is an extremely powerful piece of open source software which allows you to run a full-featured software based PBX on your computer. Typically Asterisk is run on Linux, but it has been known to run on Mac OSX and in some cases even Microsoft Windows.

Asterisk is extremely powerful and versatile, but requires dedication to get it up and running. Asterisk is NOT plug and play software; and because of its extremely versatile nature is typically difficult for first-time users to setup.

Below we have listed resources to help you in configuring Asterisk (or Asterisk@Home); as well as a basic setup guide for Asterisk. Because of the complexity of Asterisk we cannot provide detailed support in helping you get Asterisk running; but you can find help from many Asterisk experts in the resource list below.

DID based routing with Asterisk

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
Main Project Pages:
Asterisk Homepage
Asterisk@Home

Help / Support:
Asterisk Support Page
Asterisk Forum
Asterisk Wiki
Voxilla Asterisk Forum
Broadband Reports VoIP Forum

Setup Guides:
Nerd Vittles Asterisk@Home Newbie Guide
Nerd Vittles Asterisk Tutorials
Toms Networking Asterisk@Home Setup Guide
VoIP-info.org Asterisk Installation Tips


BASIC Asterisk CONFIGURATION FOR CALLCENTRIC

STEP 1 Setup trunk and global options:
Edit the sip.conf file with your favorite text editor and make the following changes:

  • Add/change the [general] section with the following parameters:
    [general]
    dtmfmode=rfc2833
    context=from-callcentric
    srvlookup=yes
    register => 1777MYCCID:[email protected]
    

  • Add the following section to define the Callcentric trunk/peer:
    [callcentric]
    type=peer
    context=from-callcentric
    host=sip.callcentric.net
    fromdomain=sip.callcentric.net
    username=1777MYCCID
    fromuser=1777MYCCID
    secret=SUPERSECRET
    insecure=very
    canreinvite=no
    disallow=all
    allow=ulaw
    

  • Add an extension to handle calls to/from your SIP phone. This is just a sample. Refer to Asterisk documentation and your SIP phone documentation for details. 123 is the extension of your phone.
    [123]
    context=to-callcentric
    type=friend
    username=123
    secret=PASSWORD
    host=dynamic
    
  • Make sure to save your changes once done.
STEP 2 Edit the file extensions.conf:
  • Add the following section to route calls FROM callcentric TO your SIP phone with extension 123:
    [from-callcentric]
    exten => s,1,Dial(SIP/123)

  • Add the following section to route calls FROM your SIP phone TO callcentric:
    [to-callcentric]
    exten => _X.,1,Dial(SIP/${EXTEN}@callcentric)
STEP 3 Verify Asterisk operations
  • Connect to asterisk console by running the following from the command line:
    asterisk -r

  • Verify that Asterisk is registered to callcentric with console command 'sip show registry'
    *CLI> sip show registry
    
    Host                    Username    Refresh	State
    sip.callcentric.net:5060    1777MYPHONE 17      Registered
    

  • Verify that your SIP phone is registered to Asterisk with console command 'sip show peers'
    pbx*CLI> sip show peers
    
    Name/username   123/123
    Host            10.11.22.33
    Dyn Nat ACL     D
    Mask            255.255.255.255
    Port            5060
    Status          Unmonitored
    

    If you see Host as "(Unspecified)" and Port as "0", then your SIP phone is not configured correctly.

  • Disconnect from Asterisk by typing exit.
STEP 4 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.