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Www.convergencetechnologycenter.org DUE 402356 Introduction to Asterisk with Vincente D’Ingianni Director of Professional Services Binary Systems, Inc.

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Presentation on theme: "Www.convergencetechnologycenter.org DUE 402356 Introduction to Asterisk with Vincente D’Ingianni Director of Professional Services Binary Systems, Inc."— Presentation transcript:

1 DUE Introduction to Asterisk with Vincente D’Ingianni Director of Professional Services Binary Systems, Inc.

2 2 What is TrixBox? TrixBox is an Asterisk distribution pre-packaged with the following additions:  CentOS - a free distribution of RedHat Linux  FreePBX – a web-based configuration tool  And many other bells and whistles. TrixBox allows easy user access to Voice Mail, Conferencing, etc. TrixBox removes many of the complications of configuring Asterisk.

3 3 TrixBox Installation See Installation Video

4 4 TrixBox Admin Login Default UserID: maint Default Password:password

5 5 TrixBox Admin Mode Allows configuration of your Asterisk system through FreePBX or configuration files.

6 6 FreePBX Module Admin Feature modules must be installed prior to configuration. Install the following modules:  Feature Code Admin  Voic  IVR  Ring Groups  Time Conditions  Music on Hold  Info Services  Backup & Restore Click “Process” when done.

7 7 Module Admin Confirmation Click “Confirm” to install.

8 8 Apply Configuration Changes All changes must be applied before they become active. Look for the Red bar at the top of the FreePBX window to signal when changes must be applied. Simply click the bar to apply the changes.

9 9 FreePBX Configuration Return to FreePBX Setup to see your new modules.

10 10 Configure SIP Extensions Add phone extensions to your PBX.

11 11 Add Extension Name & Number User Extension is the extension number. Display Name is the Caller ID Name. Secret is the SIP registration password. Leave other fields alone.

12 12 Fax, Privacy, Recording, and Voic Other features for the extension can be configured here. Click “Submit” when finished. Add other extensions as necessary. Don’t forget to click the “Apply Changes” bar.

13 13 Configure X-Lite Soft Phone Download and install the X-Lite Soft Phone from Configure your Soft Phone to match the exensions in FreePBX. Point the Soft Phone to the IP address of the Asterisk system. Configure multiple Soft Phones on other computers for each extension.

14 14 Configure X-Lite Soft Phone Download and install the X-Lite Soft Phone from Configure your Soft Phone to match the exensions in FreePBX. Point the Soft Phone to the IP address of the Asterisk system. Configure multiple Soft Phones on other computers for each extension.

15 15 X-Lite Soft Phone Making Calls Between Extensions X-Lite Soft Phone IP Network Extension 101 dials 102 to make a call on the PBX.

16 16 Simple IAX2 Trunking Create a “Virtual Trunk” between two Asterisk PBXs.

17 17 Simple IAX2 Trunking Leave the top fields blank

18 18 Simple IAX2 Trunking Put the Peer PBX IP address on the PEER Details Configure both PEER and USER type=friend There is no registration or password required between PBXs, so do not implement this on a real system unless you are sure of what you are doing.

19 19 Create an Outbound Route Set the Dial Pattern to match any dialed number string beginning with a “9” Put your IAX2 trunk in the Trunk Sequence list. Note: A dial pattern of “9|.” will strip off the 9 and pass the rest of the digits to the trunk.

20 20 Create an Inbound Route Allow inbound calls from any trunk to be routed to a specific destination. By not specifying a DID Number or Caller ID Number, any inbound call will match.

21 21 Set the Inbound Route Destination Direct the inbound route to a specific extension, ring group, IVR, etc.

22 22 General Settings Adjust the Voice Mail, Faxes, Security, etc. Allow anonymous inbound SIP Calls makes testing easy, but can be a security risk.

23 23 X-Lite Soft Phone Making Calls Between PBXs X-Lite Soft Phone IP Network Extension 101 dials 9102 to make a call across the IAX trunk between the two PBXs.

24 24 Flash Operator Panel Monitor call activity in real time on the Flash Operator Panel.

25 25 Congratulations You have now created a fully functional VoIP PBX system ideal for teaching communication fundamentals. Expand the system with real VoIP Gateways or SIP termination to the PSTN for real applications. Experiment with IVRs, Ring Groups, Time-of- Day routing, etc. Best of all, it is free!


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