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Learn about TrixBox, a pre-packaged Asterisk distribution that simplifies the configuration of your Asterisk system. This guide covers installation, configuration, and the use of features such as voice mail, conferencing, and more.
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Introduction to Asterisk with Vincente D’Ingianni Director of Professional Services Binary Systems, Inc. vincente@binary-systems.com
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.
TrixBox Installation See Installation Video
TrixBox Admin Login • Default UserID: maint • Default Password: password
TrixBox Admin Mode • Allows configuration of your Asterisk system through FreePBX or configuration files.
FreePBX Module Admin • Feature modules must be installed prior to configuration. • Install the following modules: • Feature Code Admin • VoiceMail • IVR • Ring Groups • Time Conditions • Music on Hold • Info Services • Backup & Restore • Click “Process” when done.
Module Admin Confirmation • Click “Confirm” to install.
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.
FreePBX Configuration • Return to FreePBX Setup to see your new modules.
Configure SIP Extensions • Add phone extensions to your PBX.
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.
Fax, Privacy, Recording, and VoiceMail • 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.
Configure X-Lite Soft Phone • Download and install the X-Lite Soft Phone from www.counterpath.com • 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.
Configure X-Lite Soft Phone • Download and install the X-Lite Soft Phone from www.counterpath.com • 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.
IP Network Making Calls Between Extensions X-Lite Soft Phone X-Lite Soft Phone Extension 101 dials 102 to make a call on the PBX.
Simple IAX2 Trunking • Create a “Virtual Trunk” between two Asterisk PBXs.
Simple IAX2 Trunking • Leave the top fields blank
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.
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.
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.
Set the Inbound Route Destination • Direct the inbound route to a specific extension, ring group, IVR, etc.
General Settings • Adjust the Voice Mail, Faxes, Security, etc. Allow anonymous inbound SIP Calls makes testing easy, but can be a security risk.
IP Network Making Calls Between PBXs X-Lite Soft Phone X-Lite Soft Phone Extension 101 dials 9102 to make a call across the IAX trunk between the two PBXs.
Flash Operator Panel • Monitor call activity in real time on the Flash Operator Panel.
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!