Configuring a VoIP Device / Softphone

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 and trixbox / freePBX

RESOURCES

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

BASIC Asterisk CONFIGURATION FOR CALLCENTRIC

1 Edit file sip.conf: Add/change [general] section to indicate the following parameters: [general] dtmfmode = rfc2833 context=from-callcentric srvlookup=yes register => 1777MYCCID:SUPERSECRET@callcentric.com/1777MYCCID Add the following section to handle calls to/from callcentric: [callcentric] type=peer context=from-callcentric host=callcentric.com username=1777MYCCID secret=SUPERSECRET fromuser=1777MYCCID fromdomain=callcentric.com insecure=very 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 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 => _XX,1,Dial(SIP/${EXTEN}@callcentric) 3 Verify Asterisk operations 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". 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).