From VoIP.ms Wiki
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.
On console, login into you new trixbox with the username: root and the password you selected during installation.
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)|
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.
Once there, click on Trunks option and select the trunk of your preference, SIP or IAX2 protocol based.
[For this example, we will add a SIP trunk]
Now we are in the SIP Trunk Section.
- A Dial Rule controls how calls will be dialed on this trunk. It can be used to add or remove prefixes. Numbers that don't match any patterns without a + and | to add or remove prefix.
Only the first matched rule will be excuted and the remaining rules will not be acted on.
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 Name: Gives this trunk a unique name. NOTE: For IAX trunks, the name must be set as voipms
- PEER Details: Modify the default connection parameters for VoIP.ms.
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
- User Details: Modify the default user connection parameters (leave in blank)
- Register String: Establish the connection with your voip.ms account, and determine the port used. Example:
You can select one of our different servers: Atlanta, GA: atlanta.voip.ms (22.214.171.124) Chicago, IL: chicago.voip.ms (126.96.36.199) Dallas, TX: dallas.voip.ms (188.8.131.52) Houston, TX: houston.voip.ms (184.108.40.206) Los Angeles, CA: losangeles.voip.ms (220.127.116.11) New York, NY: newyork.voip.ms (18.104.22.168) Seattle, WA: seattle.voip.ms (22.214.171.124) Tampa, FL: tampa.voip.ms (126.96.36.199) Montreal 2,QC: montreal2.voip.ms (188.8.131.52) Toronto 2, ON: toronto2.voip.ms (184.108.40.206) Montreal,QC: montreal.voip.ms (220.127.116.11) Toronto, ON: toronto.voip.ms (18.104.22.168) London, UK: london.voip.ms (22.214.171.124)
Then please submit these changes.
Don't forget to apply all these changes click on "Apply Configuration Change" and restart Asterisk on every change.
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".
- User Extension: The number assigned to 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.
- SIP Alias: If you want to support direct SIP dialing.
- 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:
10 seconds are 2 rings 15 seconds are 3 rings 20 seconds are 4 rings 25 seconds are 5 rings
- 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.
- Emergency CID: The Caller ID we always be set when dialing out and outbound route flagged as Emergency.
- DID Description: Use an internal description for your extension. Example: Office, Fax, etc...
- Add Inbound DID: If you have a directly DID associated to this extension please put this here.
- Secret: Please set the password that you want for this extension, you will use this password when you setup this extension with a softphone or a device.
- dtmfmode: By default is rfc2833.
- Record Incoming: You can record all the incoming calls for this extension.
- Record Outgoing: You can record all the outgoing calls for this extension.
Voicemail & Directory
- Voicemail Password: This is the password used to access the Voicemail system.
- Email Address: The email address that Voicemail are sent to.
- Pager Email Address: Email address that shorts Voicemail notifications are sent to.
- Email Attachment: Option to attache Voicemail to mail.
- Play CID: Reads back the callers telephone number.
- Play Envelope: Envelope control whether or not the Voicemail system will play the message envelope.
- Delete Voicemail: If set yes the messages will be deleted from the voicemail box (after having been emailed).
- VM Options: Separate option with pipe (|)
The outgoing calls are sent over trunks, and determined by the configuration of the Outbound Routing page.
- Go to outbound Routes section >> add route.
- Route Name: The name of the route, should be use to describe what type of calls this route matches Ex, Local or International.
- Route Password: A route can prompt a user for a password before do his calls. (optional)
- PIN Set: If use this option leave the password option in blank.
- Emergency Dialing: Select this option if this set of routes is used for emergency calls.
- Intra Company Route: Selecting this route will treat this route as intra company route.
- Music On Hold: Default (optional)
- Dial Patterns: A dial pattern is a unique set of digits that will that will select this trunk, if you want to add a outbound rules please use the "Dial patterns wizards":
Local 7 digits: NXXXXXX Local 10 digits: NXXNXXXXXX Information: 511 411 311 811 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.
Now save the changes.
To learn more about how the dialing rules and patterns work, please refer to the article Dialing Rules and Patterns
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.
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.
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.
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.
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.