From VoIP.ms Wiki
In this entry we show how to get started in Trixbox, the initial settings and first steps.
We talk a little about what is Trixbox and what we can do with it.
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.
With Trixbox we are many features including:
- Creating extensions
- Sending voice messages to e-mail
- Conference Calls
- Digital Receptionist (IVR)
- Calling Queues
Once installed Trixbox and once they load, we going to be in this section that asks us to connect us with the server. Will ask us for a username and password, by default is: Login: root / Password: password (or select one if you set at the time of installation).
|After that please select the IP that are show it.||And paste this on your browser.|
|Once we're there, we will log by clik in "Switch"||Please log in, by default the user/password are: maint/password.|
Now we are in the main Trixbox interface, here is where we will see an overview of our system. To begin configuring your Trixbox with our VoIP.ms account go to the PBX section >> PBX Settings.
Once you get in, please go to the trunk section and then please select the trunk of your choice, SIP or IAX2.
Now we are in the SIP Trunk Section.
- Outbound Caller ID: Format "caller name" <5094623XX> You can aslo use the magic string hiden to hidde the Caller ID sent out over Digital Lines.
- 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 Ex: VoIP.ms
- PEER Details: Modify the default connection parameters for VoIP.ms, you may need to add here the following trunk with your account information:
canreinvite=nonat ; nat=yes ; uncomment if behind nat context=from-trunk host=atlanta.voip.ms secret=johnspassword ;your password type=friend username=100000 ;your account disallow=all allow=ulaw ; allow=g729 ; uncomment if you purchased g.729 from Digium fromuser=100000 ;your account 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 wit you voip.ms account Example:
You can select one of our different servers: Atlanta, GA: atlanta.voip.ms (188.8.131.52) Chicago, IL: chicago.voip.ms (184.108.40.206) Dallas, TX: dallas.voip.ms (220.127.116.11) Houston, TX: houston.voip.ms (18.104.22.168) Los Angeles, CA: losangeles.voip.ms (22.214.171.124) New York, NY: newyork.voip.ms (126.96.36.199) Seattle, WA: seattle.voip.ms (188.8.131.52) Tampa, FL: tampa.voip.ms (184.108.40.206) Montreal 2,QC: montreal2.voip.ms (220.127.116.11) Toronto 2, ON: toronto2.voip.ms (18.104.22.168) Montreal,QC: montreal.voip.ms (22.214.171.124) Toronto, ON: toronto.voip.ms (126.96.36.199) London, UK: london.voip.ms (188.8.131.52)
Then please submit these changes.
Remember to apply all these changes click on "Apply Configuration Change"
Now let's create an extension,these are very important, as they are an important peace to make and receive calls.
Just on the left we have all these options, please go to Basic >> Extensions and once we are there goes to "Add Extension". And then you can select your device type.
So then we gonna start to configure our first extension.
- User Extension: The extension number to dial to reach this number.
- Display Name: The caller ID name for calls from this user.
- CID Num Alias: This is like your 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 formta: "caller name" <###########>.
- Ring Time: Numbers of ring before goes to the voicemail. You can put the number in seconds please note that format:
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 put the password that you want to set 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 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 (|)
Save those changes.
Outgoing calls are sent over trunks as determined by the configuration of the Outbound Routing page. Please go after all make sure to save your changes by clicking on the "Apply Configuration Changes" link, so then go to the Outbound Routes section >> add route.
- Route Name: The name of this 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 Toll Free: 1800NXXXXXX 1888NXXXXXX 1877NXXXXXX 1866NXXXXXX Long Distance: 1NXXNXXXXXX International: 011 Information: 411 311 Emergency: 911
- Trunk Sequence: The trunk sequence control the orders of trunks that will be used when the above dial patterns are matched. Ex: You can select the trunk that will create a few moments ago.
Now save those changes.
Using the system
Now we have done all the configuration to start to make calls, we can setup our extensions with a Softphone and devices.
In this case we gonna make some changes about the configuration:
- At this moment our SIP user name will be the number of our extension. EX: 235
- The password will be the one you set with your extension.
- And out server will be the IP address of the Trixbox Main Portal:
So then we have a few information to setup our service:
- SIP User Name: 235 (your extension number)
- Domain : 192.168.41.111 (IP address from the Trixbox Main Portal)
- Password: the password you set in your extension.
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 .