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Supported devices |
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| Asterisk |
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| GENERAL INFORMATION |
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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
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| RESOURCES |
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Main Project Pages:
Asterisk - http://www.asterisk.org
Asterisk@Home - http://asteriskathome.sourceforge.net
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 |
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| BASIC Asterisk CONFIGURATION FOR CALLCENTRIC |
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| 1 |
Edit file sip.conf: |
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- Add/change [general] section to indicate the following parameters:
[general]
dtmfmode = rfc2833
context=from-callcentric
srvlookup=yes
register => 1777MYCCID:SUPERSECRET@callcentric.com/1777MYCCID
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas
- Add the following section to handle calls to/from callcentric:
[callcentric]
type=peer
context=from-callcentric
host=callcentric.com
defaultuser=1777MYCCID
secret=SUPERSECRET
fromuser=1777MYCCID
fromdomain=callcentric.com
insecure=port,invite
- Add a section 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=PHONESECRET
host=dynamic
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| 2 |
Edit the file extensions.conf: |
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- 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 => _XX,1,Dial(SIP/${EXTEN}@callcentric)
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| 3 |
Verify Asterisk operations |
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- Connect to asterisk console by running:
# asterisk -r
- Verify that Asterisk is registered to callcentric with console command 'sip show registry'
*CLI> sip show registry
| Host | Username | Refresh State |
| callcentric.com: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".
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| 4 |
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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