Trixbox

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Trixbox is currently in an abandoned state. No changes have been published since June 2010. Customers should look for alternatives.

Trixbox is an operating system distribution that has the distinction of being a telephone exchange (PBX) software based on open source Asterisk PBX. Like any PBX, to interconnect a company's internal phone and connect the telephone network.


Contents

First Steps

Trixbox.jpg

On console, login into you new trixbox with the username: root and the password you selected during installation.

Trixbox log.jpg IP Browse.jpg

When you log in the system will tell you what IP address it received from your DHCP server. You can give the system a permanent address now by typing system-config-network or setting the IP address from the GUI. If you reconfigured the IP address, restart the network device by running service network restart. To continue configuration connect to your system with a web browser using the assigned IP you specified in the previous step.


Once we're there, we will log by clicking "Switch" Please log in, by default the user/password are: maint/password.(We strongly recommend to change the default username and password for security reasons)
Switch 2.jpg Switch.jpg

Basic

Trunk Configuration

Now, we finally should see the Trixbox GUI Web interface, to start the Tribox configuration, look for PBX section >> PBX Settings, in the left menu.

Trixbox inter.jpg

Once there, click on Trunks option and select the trunk of your preference, SIP or IAX2 protocol based.

Trunks 1.jpg

[For this example, we will add a SIP trunk]

Now we are in the SIP Trunk Section.

SIP trunk 1.jpg

Only the first matched rule will be excuted and the remaining rules will not be acted on.

Rules:

   X mathes any digit from 0-9
   Z matches any digit from 1-9
   N matches any digit from 2-9

A few samples :

   A local 7 digts number: NXXXXX 
   Toll-Free numbers : 1800NXXXXX
   There are a few samples , you can set all your rules in the outbound routes section. 


Trunk Out.jpg

   canreinvite=nonat
   ; nat=yes ; uncomment if behind nat
   context=from-trunk
   host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location)
   secret=***** (password associated with the Main or Sub-account)
   type=peer
   username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
   disallow=all
   allow=ulaw
   ; allow=g729 ; uncomment if you purchased g.729 from Digium
   fromuser=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
   trustrpid=yes
   sendrpid=yes
   insecure=port,invite
   qualify=yes

Register String.jpg

      i.e, 123456:password@atlanta.voip.ms:5060 

You can select one of our different servers, for example atlanta.voip.ms. Please check the complete list at FAQ page :[[1]]

Then please submit these changes.

Don't forget to apply all these changes click on "Apply Configuration Change" and restart Asterisk on every change.

Config change.jpg

Extensions

Now let's create an extension,this is important, you may want to have at least one extension set, in order to route your incoming calls to it, and place outgoing calls.

Please go to Basic Menu >> Extensions option >> Once there go to "Add Extension".

Extension1.jpg

Ext2.jpg

Extension Options

   10 seconds are 2 rings 
   15 seconds are 3 rings
   20 seconds are 4 rings 
   25 seconds are 5 rings 

Assigned DID/CID

Device Options

Exten 3.jpg

Recording Option

Voicemail & Directory

   review=yes|maxmessages=60

Save changes.

Outbound Routes

The outgoing calls are sent over trunks, and determined by the configuration of the Outbound Routing page.

Out Route.jpg


   Local 7 digits: NXXXXXX
   Local 10 digits: NXXNXXXXXX
    Information: 511
                411
                311
                811
   Emergency: 911 
   Echo test: 4443
   DTMF test: 4747

Now save the changes.

To learn more about how the dialing rules and patterns work, please refer to the article Dialing Rules and Patterns

Inbound Route

Information

The 'Inbound Routes' page lets you configure the destination that incoming calls will use. When a call is recieved by Asterisk from a trunk, the DID and/or Caller ID is matched and the call is dispached as per your settings.

Trixbox-inbound-route.png

DID Number This is usually your DID Number, putting that in here will match calls coming from that number. Leaving this blank will match 'any'.

CID Number The Caller ID number sent to your machine. This is not something you should trust, as it is easily spoofable (both with Voice over IP and normal telephone lines). Leaving it blank will, again, match any.

You can leave both of these blank to match any call, from any caller.

Fax Handling

Although the Trixbox has the ability to handle Faxing, please remember that this service is not officially supported with us, some of our users have confirm this works using g711u codec and Premium route, however remember this service is not guaranteed.

Privacy Manager Turn this on to ask for the callers Caller ID if not provided.This is useful for telemarketers, as they are loathe to divulge this information and will usually hang up.

Options

Pause After Answer The number of seconds we should wait after performing an Immediate Answer. The primary purpose of this is to pause and listen for a fax tone before allowing the call to proceed.:

Alert Info ALERT_INFO can be used for distinctive ring with certain SIP devices. The standard names are 'Bellcore-dr1' to 'Bellcore-dr7', Snomphones can additionaly use a http:// url of a WAV or MP3 file.

Set Destination This is a standard destination option group.

Security Measures

We strongly recommend you to change the password in your account, PBX system and extensions on it, periodically

As a preventive measure you also can disable International calls on your account. From the customer portal >> Main Menu >> Account Settings >> Account Restrictions. These settings define the restrictions the system will use when you place calls to either USA48, Canada or International Numbers.

We strongly recommend to specifically select only the countries that you on your regular traffic (outgoing calls). You can do this by clicking in Currently Allowed: All Countries Allowed >> Click here to manage list of allowed countries

Additionally, you can use in your SIP.Conf alwaysauthreject = yes, what the alwaysauthreject parameter does when set to yes, is it will ALWAYS return an authentication error instead of a "404 - Not Found", even when the extension doesn't exist. This mess up scanners, because the program detect an "existing" extension even if it's not present on the server. Unfortunately, it's way from being fool proof, but it's a nice security addition that you can set to your Asterisk based PBX.

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