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Trixbox v 2.0 with VarPhonex Trunk
The purpose of this document is to provide a step by step installation guide of Trixbox using VarPhonex as the VoIP provider. We have only included the information we considered was necessary to perform a basic installation. You can always visit http://asteriskathome.sourceforge.net/ in order to learn about all the features provided as well as additional information you may need. Just follow this installation guide step by step and you will be able to install and configure your Trixbox with our VarPhonex service in no time!
Table of Contents What is Asterisk? 1 Pre-Installation Tasks 2 Installation 3 Securing your Trixbox server 4 Using FREEPBX to configure your Trixbox server 5 Other Tasks 6 Routing and managing multiple DIDs
Who can use Trixbox? Trixbox can be configured in different ways according to your needs.
Asterisk is an open source PBX that allows regular and sip phones to communicate with each other. Like any PBX system, Asterisk has features such as: Voicemail, conferencing, call distribution. Trixbox is an iso image of a pre-configured Asterisk server which makes installation and deployment easier. Trixbox contains a full version of Asterisk and other pre-configured applications considered add-ons. After installing Trixbox, you will have a fully functional PBX which can be customized according to your needs.
1.1 Meet the minimum or recommended hardware requirements The faster the system you use to run Asterisk, the more simultaneous calls it will be able to handle. A 500MHz PIII with 128 Megs of RAM should easily meet the needs of the average home use. 2Gb Hard Disk minimum. Keep in mind that these are the minimum requirements. If you are planning to use Asterisk in an office environment where voicemail and call monitoring will be used, we would suggest you use a PIV CPU, at least 512 MB of RAM and at least a 40 GB hard drive.
Download the latest .ISO from http://www.trixbox.org/downloads and burn it to a CD. One program you can use for this Alcohol 120% located at:
1.3 Set up your router/firewall so Trixbox can communicate with VarPhonex via SIP through NAT For Trixbox to communicate successfully with VarPhonex using SIP through a NAT, you have to make sure your router/firewall forwards the following ports to your LAN/Private IP address assigned to the Trixbox server. Be sure the LAN/Private address is statically assigned to the Trixbox server and it is not assigned dynamically via DHCP. In your firewall’s configuration forward the following ports to your Trixbox’s IP address:
Note: We do not support IAX or IAX2. We included them in the table as a reference.
1.4 Setup for changing (dynamic) Internet IP address Most ISPs do not provide a “private static IP address” which would be recommended to run Trixbox. The average ISP provides Dynamic (DHCP) addresses which makes it a little more difficult for users to run Trixbox. The work around for this problem is “Dynamic DNS”.
What is dynamic DNS? Dynamic DNS allows an internet domain name to be assigned to a dynamic IP address. Some dynamic DNS providers provide a piece of software that can be installed in the server. This software works in the background and it tracks any change in the IP address and sends it to their database. This way the domain name will be always updated with the correct IP address as soon as it changes. There are some routers in the market that have this feature built in which makes unnecessary to install any software in the server. All you have to do is get an account with the provider and configure it in the router.
How do I use Dynamic DNS with Trixbox? You need to edit the sip.conf file. Inside of FREEPBX, click Maintenance ----> Config Edit ----> sip_nat.conf. Inside of sip_nat.conf add the following and click "Update":
To determine your local NETWORK address (NOT the IP address!!) you have to know a little about your subnet mask (255.255.255.0 numbers).
If you are using NAT enter the following:
The [general] context of your sip.conf file should look like this:
Insert the CD you created using the ISO image and make sure that your Bios is configured to boot from a CD-ROM or DVD-ROM. Boot the computer and press ENTER when prompted. This will erase all the information on the hard drive and install your Trixbox. Once your Trixbox server is installed, it will have all the applications and the operating system itself with default passwords; That is why it is recommended that you unplug your server from the network in order to avoid any hacker attack. After Linux has loaded, the CD will eject. Remove the CD from the system and wait for the system to reboot. Booting the system might take a while, depending on the speed of your computer. Once this process is complete, log in to your new Trixbox system with the user = root and the password you created during the installation.
3 Securing your Trixbox server 3.1 Configure your Trixbox server with a static IP address In order to change the default passwords, we need to assign your Trixbox a static IP address. At the CentOS command line type:
3.2 Changing your default FOP password The default password for the Flash Operator Panel is: Note that 0 is a “zero”
3.3 Changing your default meetme password To change the default type the following into the CentOS command prompt:
3.4 Changing your default System Mail password To change the default password type the following into the CentOS command prompt:
3.5 Changing your default Sugar CRM Password Access SugarCRM from your web page by typing HTTP://YourAsteriskIPaddressHere into your web browser. The default login and password are:
3.6 Updating patches to CentOS It is recommended that you install CentOP patches. From the CentOS command line, run the following command:
4 Using FREEPBX to configure your Trixbox server Asterisk Management Portal makes Asterisk configuration easier by providing a graphical method (through a web browser). FREEPBX allow you con configure the textual configuration files that Asterisk needs to function. FREEPBX can configure the following in asterisk: For VarPhonex configuration purposes we will need to enable some of the modules in FreePBX
5. From the device drop down menu select “Generic SIP device” and click submit.
Example
NOTE: If the extension you are configuring will connect remotely (outside the Local Area Network) you will need to change the NAT option to yes. Just create the extension, submit the changes and go back to edit it. You will see NAT=never; change it to NAT=yes Every time you make a configuration change and click “Submit” a RED bar will appear at the top of the screen “Apply Configuration Changes”. This bar will reload the . conf files. Click this bar in order for the changes to take effect.
4.3 Configuring trunk for inbound and outbound calls
4.4 Configuring Outbound Routing You will need to allow calls from your phones to go out on a specific trunk. When having more than one trunk, you will need to setup dialing rules (dialing patterns) in order to specify which calls should go out on which trunk.
4.5 Configuring Inbound Routes NOTE: YOU WILL NOT BE ABLE TO RECEIVE CALLS IF YOU DO NOT CONFIGURE AT LEAST ONE INBOUND ROUTE Configuring inbound routes will allow calls from VarPhonex go someplace in your PBX. Using FREEPBX
System Recording will allow you to record your own voice prompts or create one putting several built-in voice prompt files together to create the one you need. For this example will use the “Built-in Recordings” option to create an IVR that will play “Welcome, please enter the extension number. Thank you for calling”. Using FreePBX
6. After saving all files your recording will be created with the name of the first file selected. In this case, “welcome.”
4.7 IVR (Digital Receptionist) You use the Digital Receptionist to make IVR's, Interactive Voice Response systems. When creating a menu option, apart from the standard options of 0-9,* and #, you can also use 'i' and ’t’ destinations. 'i' is used when the caller pushes an invalid button, and 't' is used when there is no response. If those options aren't supplied, the default ’t’ is to replay the menu three times and then hang up, and the default 'i' is to say 'Invalid option, please try again' and replay the menu. After three invalid attempts, the line is hung up.
Using FreePBX
All incoming calls will be routed to the “welcome” system recording allowing callers to select the desired extension.
Up to this point we have performed a basic installation and configuration of Trixbox.
NOTE: THE STANDARD INSTALLATION OF TRIXBOX DOES NOT COME WITH G723 AND G729 CODECS. IF YOU RESTRICT YOUR TRUNK TO ONLY USE THESE AND THEY ARE NOT INSTALLED YOU WILL NOT BE ABLE TO PLACE CALLS.
5.1 Install low bandwidth codecs You can find the specific codes for your type of CPU in the following link: DISCLAIMER: You might have to pay royalty fees to the G.729/723 patent holders for using their algorithm. To install the codec move .so file into /usr/lib/asterisk/modules directory in your Asterisk server.It is very important that you choose the codec according to the CPU you server has. In case you choose the wrong type Asterisk will not load and give you an error message. All you have to do is remove the file and restart your server. Here is the command to remove files in CentOS (Linux): rm filename (replace “filename” with the name of the codec file) Once you determine the right file for your server, enter the following commands in your server’s prompt and press enter Assuming that the file I need is codec_g729-gcc-pentium4-no-sse.so Enter the command: Assuming that the file I need is /codec_g723-gcc-pentium4-no-sse.so wget http://asterisk.hosting.lv/built-for-asterisk-1.2/ codec_g723-gcc-pentium4-no-sse.so In order to determine if we downloaded the correct file, run the following commands: asterisk –r [press enter] If the file was loaded correctly, you will see the translations under G729. Perform the same operation to install the G723 codec. 5.2 Restrict the VarPhonex trunk to the above mentioned codecs. Using FreePBX
5.3 Restrict Asterisk to use low bandwidth codecs for remote extensions. Use a pc on your network that has a web browser and connect to your Trixbox box using HTTP://PutYourTrixboxIpaddressHere.
6 Routing and managing multiple DIDs There will be times where you will need to point different DIDs to different contexts, IVRs (voice prompts) or extensions to accomplish the configuration you need. For example, you may need to provide two different numbers to your customers; one for English and one for Spanish. You may simply want to configure one DID as the main number but provide each extension in your Trixbox its own DID that will work as its direct number. You can even configure two different companies in the same Trixbox and provide a DID for each of them. This is what you will need to do if you want to configure VarPhonex DIDs: Note: If the DID you want to configure is assigned to a virtual number that already has a registration string in the sip_nat.conf file; you will need to delete it, save your changes and wait until the registration has expired in our sip proxy. You can see this registration in the control panel under the “Virtual Numbers” screen.
Once you verify that the VN was registered in control panel, Configure an inbound route based on the DID.
I cannot receive calls. Check the following:
I can receive calls but I cannot make any.
The Virtual number configured in my Trixbox does not show registered in sip.varphonex.com
My calls do not have good quality.
When I place calls my caller ID does not show correctly. When you place IP calls the system will send Toronto, ON and override any name that appears in Account Info (Control Panel settings) for the Virtual Number. This is sent on any call placed via a VN that is not associated with a purchased DID. When you place PSTN calls via your 7 digit Virtual Number that is not associated to an incoming purchased DID number, the Caller ID will display 6477233283 as the number the call was placed from or it may show 588 + your 7 digit virtual number. When you place PSTN calls via a purchased incoming DID number the DID will display. The system will send the display name (from account info in your control panel) as well (PSTN call via DID) but we cannot guarantee the PSTN provider will pass it down. Again, the name sent is not guaranteed. We do transmit the name from the Account Info, however, not all carriers support our transmission of that data and may rely instead on 3rd parties for the name information. There are also areas where the transmission of caller id is not supported by the carrier. It is also rare for the name part of caller ID to show up on any cell phone. Caller ID cannot be blocked.
Caller ID cannot be manipulated in your control panel or by a calling device. We have seen cases where Caller ID is sent as unknown, which is resolved when the soft phone is completely uninstalled and reinstalled or in the case of a hard phone, is reset to manufacturers default settings and reconfigured. We can not guarantee Caller ID. If the callees phone number Provider looks to associate the DID with a Name, the CID will probably be displayed as Unknown, No Data, or just blank.
Notice regarding support: In an effort to provide the best possible support, we offer the following options to Trixbox/Asterisk users:
These are the only support options that we provide at this time.
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