Asterisk IP PBX Learning Tutorials - Part 10
Asterisk Video Tutorial - Trixbox Features – Part 2
Trixbox features’ overview Part 2
In this video I will show you how to administer you trixbox system
In order to get to trixbox administration page you have to click on “switch” link near the “User mode” status text. The default username for every trixbox installation is “maint” and password is a word “password” in lowercase.
Once you are in administration mode, you will be presented with status overview of your system. Here you can see “System Statistics”, “FreePBX statistics”, freepbx is the name of the webinterface that trixbox use, so it will be mention on pages a lot. Also you can check your system uptime and “server status” to be sure that asterisk and other essential software is running ok.
As you can see using trixbox’ administration mode you can configure a huge number of options for your asterisk system without a need to memorize them all. I will not cover all of them, but you can get help on any single feature and option just by moving your mouse over it, as you can see the system is very well documented so you can be sure you won’t get lost.
Lets take a look at important configuration options. Extensions is obviously is one of them. Clicking on extension will take you to a page were you can see a list of all the extensions already configured on your system and also an option to create new one.
In order to create new extension you have to choose what technology it will use and then click “Submit” button. Here you can define a wide range of options, and as I told you before has a help in case you are not sure about you need it or not. After you are done with setting all needed options you have to click submit button again.
You have to remember one thing, while you are doing any changes through web interface, you are changing that in memory, meaning in order to get them applied to asterisk configuration and reload it you have to click on “apply configuration changes” orange ribbon in the upper part of the page. It will ask you for confirmation and if you are sure then your changes will be applied to asterisk.
Another important option you need to setup is trunks. Here you define your VoIP provider with whom you will be able to place calls to. As with extensions, first thing you have to do is to define what technology your provider is using. As usual, every option is very well documented and necessary options are placed in configuration, so you can be sure that you will not miss anything. Moreover, most of the providers have a help page on their websites, where you can find what options you have to exactly setup for your trixbox and asterisk in order to work with them. It is almost incredible how trixbox became almost standard and everybody supports it.
Once you are done with setting up your trunks, you have to define how your calls will be placed to and received from those provides using defined trunks. To do this you have to setup “Outbound Routes” and “Inbound Routes” respectively. Lets start with outbound ones. You have to define route name and optionally route password, if you do, every time somebody on your asterisk system try to dial out through this provider will be asked for password. Next important option is “Dial patterns” it defines how you and your users will access that provider and for what purpose. As you can see you can write here any rule you want, but there are some predefined patterns that may save your time figuring out what and how for you will use this route. For example lets assume that we will use this route for long distance calls and international ones, as you see system choose the correct settings for us. The last option you have to set while defining outbound route is what trunk we will use. Lets choose our newly created trunk. And you are done, now every time you will be placing international or long distance calls, you will be directed by system to our trunk.
Lets move on and configure inbound route, so outside users can reach those who are inside. Lets give it some distinguishing name, and the important thing here is to match the inbound number that has been dialled, this is usually number you get from your VoIP provider, and in terms of asterisk it is called DID or direct inward dial. After this you have to decide what to do with your call once it reached the system, you can set this at “Set destination” option. Here you can terminate the call with some number of reasons, connect it directly to some extension you have setuped before, redirect it to voicemail of some extension, put it in a ring group and join a conference. Putting in “ring group” means that all local extensions that are members of particular ring group will ring at once, as soon as new call will arrive. After you are done, don’t forget that you have to apply changes to you asterisk system by clicking on orange ribbon bar and choosing “Continue with reload”. There are a lot of other options that you can setup and enable through this webinterface, you can experiment with them freely and be sure that every option is very well documented.
In addition to configuration options in administration mode, you can get number of reports on system usage, like who was calling whom, what was the duration, and also some nice graphs on call distribution and other important aspects of your asterisk system.
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