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 versions 1.4 and 1.2 click here
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GENERAL INFORMATION |
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Asterisk is an extremely powerful piece of open source software which allows you to run a full-featured software based PBX on your computer.
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; 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.
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RESOURCES |
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Websites
Asterisk Homepage
Help / Support:
Asterisk Support Page
Asterisk Forum
Asterisk Wiki
Voxilla Asterisk Forum
Broadband Reports VoIP Forum
Digium Community
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Configuring the Asterisk 1.6 |
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The instructions below are meant to assist you with the basic configuration of asterisk. If you wish to perform more complicated configurations please view the asterisk documentation above and the asterisk forums.
We recommend that you read each step through in its entirety before performing the action 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" you should see an output similar to the following:
Asterisk 11.4.0 built by root @ server on a i686 running Linux on 2013-05-31 09:46:40 UTC
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STEP 1 |
Setup trunk and global options |
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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]
session-timers=refuse
- Add the following section to define the Callcentric trunk/peer:
[callcentric]
type=peer
context=from-callcentric
host=sip.callcentric.net
fromdomain=sip.callcentric.net
defaultuser=1777MYCCID
fromuser=1777MYCCID
secret=SUPERSECRET
insecure=port,invite
canreinvite=no
videosupport=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
defaultuser=123
secret=PASSWORD
host=dynamic
- Make sure to save your changes once done.
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STEP 2 |
Configuring inbound call handling |
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Edit the extensions.conf file with your favorite text editor and make the following changes:
- 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)
- Make sure to save your changes once done. If necessary restart, or start the asterisk server using the proper command, /etc/init.d/asterisk restart, restart asterisk... etc.
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STEP 3 |
Connect to your Asterisk PBX and verify connections |
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Use the IP address or hostname for your PBX system along with 123 (the extension created earlier which is the username) and the password for the 123 extension to connect to your PBX system.
- Connect to the asterisk console by running the following from the command line:
asterisk -r
- Verify that Asterisk is registered to Callcentric with the console command 'sip show registry'
*CLI> sip show registry
Host Username Refresh State
sip.callcentric.net:5060 1777MYCCID 17 Registered
- Verify that your SIP phone is registered to Asterisk with the 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. Asterisk will remain running in this case.
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STEP 4 |
That's it! You can now make a phone call |
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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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