Trixbox - VoIP.ms Wiki

Trixbox

From VoIP.ms Wiki

(Difference between revisions)
Jump to: navigation, search
[draft revision][quality revision]
(Outbound Routes)
(Basic)
 
(33 intermediate revisions not shown)
Line 1: Line 1:
 +
'''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.
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.
Line 25: Line 27:
{|
{|
|style="width:500px" | Once we're there, we will log by clicking '''"Switch"'''  
|style="width:500px" | Once we're there, we will log by clicking '''"Switch"'''  
-
|style="width:500px" | Please log in, by default the user/password are:''' maint/password.'''  
+
|style="width:500px" | Please log in, by default the user/password are:''' maint/password.'''(We strongly recommend to change the default username and password for security reasons) 
|-
|-
| [[File:Switch 2.jpg|400px]]
| [[File:Switch 2.jpg|400px]]
Line 65: Line 67:
-
[[File:Trunk Out.jpg|600px]]
+
[[File:Trunk_Out.jpg]]
* '''Trunk Name''': Gives this trunk a unique name. '''NOTE: For IAX trunks, the name must be set as voipms'''  
* '''Trunk Name''': Gives this trunk a unique name. '''NOTE: For IAX trunks, the name must be set as voipms'''  
Line 73: Line 75:
     ; nat=yes ; uncomment if behind nat
     ; nat=yes ; uncomment if behind nat
     context=from-trunk
     context=from-trunk
-
     host=atlanta.voip.ms
+
     host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location)
-
     secret=johnspassword ;your password
+
     secret=***** (password associated with the Main or Sub-account)
-
     type=friend
+
     type=peer
-
     username=100000 ;your account
+
     username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
     disallow=all
     disallow=all
     allow=ulaw
     allow=ulaw
     ; allow=g729 ; uncomment if you purchased g.729 from Digium
     ; allow=g729 ; uncomment if you purchased g.729 from Digium
-
     fromuser=100000 ;your account
+
     fromuser=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
     trustrpid=yes
     trustrpid=yes
     sendrpid=yes
     sendrpid=yes
Line 91: Line 93:
* '''Register String''': Establish the connection with your voip.ms account, and determine the port used. Example:  
* '''Register String''': Establish the connection with your voip.ms account, and determine the port used. Example:  
-
       '''your account:password@server.voip.ms:5060'''  
+
       '''i.e, 123456:password@atlanta.voip.ms:5060'''  
-
    You can select one of our different servers:
+
You can select one of our different servers, for example atlanta.voip.ms. Please check the complete list at FAQ page :[[http://wiki.voip.ms/article/FAQ#What_are_the_IP_addresses_of_VoIP.ms.C2.B4_servers_.3F]]
-
    Atlanta, GA: atlanta.voip.ms       (174.34.146.162)
+
-
    Chicago, IL: chicago.voip.ms        (64.120.22.242)
+
-
    Dallas, TX: dallas.voip.ms         (74.54.54.178)
+
-
    Houston, TX: houston.voip.ms        (209.62.1.2)
+
-
    Los Angeles, CA: losangeles.voip.ms (67.215.241.250)
+
-
    New York, NY: newyork.voip.ms      (74.63.41.218)
+
-
    Seattle, WA: seattle.voip.ms        (69.147.236.82)
+
-
    Tampa, FL: tampa.voip.ms            (68.233.226.97)
+
-
    Montreal 2,QC: montreal2.voip.ms    (174.142.75.171)
+
-
    Toronto 2, ON: toronto2.voip.ms    (174.137.63.206)
+
-
    Montreal,QC: montreal.voip.ms      (67.205.74.164)
+
-
    Toronto, ON: toronto.voip.ms        (174.137.63.206)
+
-
    London, UK: london.voip.ms         (78.129.153.20)
+
Then please submit these changes.  
Then please submit these changes.  
Line 124: Line 113:
[[File:Ext2.jpg|600px]]
[[File:Ext2.jpg|600px]]
-
* '''User Extension''': The number assign to this extension.  
+
* '''User Extension''': The number assigned to this extension.
-
* '''Display Name''': The caller ID name for calls from this extension.
+
* '''Display Name''': [[Caller ID]] name for calls from this extension.
-
* '''CID Num Alias''': This is an Alias for the Caller ID but only for your internal calls.  
+
* '''CID Num Alias''': This is an Alias for the [[Caller ID]] but only for your internal calls.  
* '''SIP Alias''': If you want to support direct SIP dialing.  
* '''SIP Alias''': If you want to support direct SIP dialing.  
'''Extension Options'''
'''Extension Options'''
-
* '''Outbound CID''': Override the caller ID when you call out. Caller ID format: "caller name" <###########>.  
+
* '''Outbound CID''': Override the [[Caller ID]] when you call out. [[Caller ID]] format: "caller name" <###########>.  
* '''Ring Time''': Number of rings before the Voicemail kick in. You can set this value in seconds please note:  
* '''Ring Time''': Number of rings before the Voicemail kick in. You can set this value in seconds please note:  
   
   
Line 141: Line 130:
* '''Call Waiting''': Set the initial call waiting state for this extensions.  
* '''Call Waiting''': Set the initial call waiting state for this extensions.  
* '''Call Screening''': Call Screening requires external callers to say their name, which will be played back to the user and allow the user to accept or reject the call.  
* '''Call Screening''': Call Screening requires external callers to say their name, which will be played back to the user and allow the user to accept or reject the call.  
-
* '''Emergency CID''': The caller ID we always be set when dialing out and outbound route flagged as Emergency.  
+
* '''Emergency CID''': The [[Caller ID]] we always be set when dialing out and outbound route flagged as Emergency.  
'''Assigned DID/CID'''
'''Assigned DID/CID'''
Line 193: Line 182:
     Local 7 digits: NXXXXXX
     Local 7 digits: NXXXXXX
     Local 10 digits: NXXNXXXXXX
     Local 10 digits: NXXNXXXXXX
-
    Toll Free: 1800NXXXXXX
+
    Information: 511
-
              1888NXXXXXX
+
                411
-
              1877NXXXXXX
+
-
              1866NXXXXXX
+
-
    Long Distance: 1NXXNXXXXXX
+
-
    International: 011
+
-
    Information: 411
+
                 311
                 311
 +
                811
     Emergency: 911  
     Emergency: 911  
-
 
+
    Echo test: 4443
 +
    DTMF test: 4747
*'''Trunk Sequence''': The trunk sequence control the order on how the trunks will be used when the dial patterns are matched.
*'''Trunk Sequence''': The trunk sequence control the order on how the trunks will be used when the dial patterns are matched.
Line 208: Line 194:
Now save the changes.
Now save the changes.
-
==Using the system==
+
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.
 +
 
 +
[[File: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'''
-
Now we have done all the configuration to start to make calls, we can setup our extensions with a Softphone and devices.  
+
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.
-
In this case we gonna make some changes about the configuration:
+
'''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.
-
* At this moment our SIP user name will be the number of our extension. EX: 235
+
'''Options'''
-
* The password will be the one you set with your extension.
+
-
* And out server will be the IP address of the Trixbox Main Portal:
+
-
[[File:Trixbox log.jpg|400px]]
+
'''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.:
-
So then we have a few information to setup our service:  
+
'''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.
-
*'''SIP User Name''': 235 (your extension number)
+
'''Set Destination'''
-
*'''Domain''' : 192.168.41.111 (IP address from the Trixbox Main Portal)
+
This is a standard destination option group.
-
*'''Password''': the password you set in your extension.
+
 +
==Security Measures==
-
If you need some help configuring your device, you can visit our wiki articles [[Softphones]] and [[Devices]]
+
'''We strongly recommend you to change the password in your account, PBX system and extensions on it, periodically'''
-
==Logout==
+
'''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.'''
-
To shut down or Trixbox, you can close your browser, and then to stop the server we go to the Trixbox CE and we put the command: '''logout''' .
+
''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''
-
[[File:Logout.jpg|400px]]
+
'''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.'''
 +
[[Category:PBXes]]

Latest revision as of 19:07, 4 December 2013

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.

Personal tools
Namespaces
Variants
Actions
VoIP.ms Wiki
VoIP.ms Blog
Configuration
Guides (English)
Guides (Français)
Guías (Español)
Toolbox