Tag Archives: trixbox

Monitoring your Peers (Asterisk extensions) and Trunks

As an admin for a telephone system, possibly one of the most useful things you can do is monitoring your peers and trunks. There are 2 great reasons you should do so:

1. You can respond to and resolve issues with your system before your users know about it, and you can be in the know if someone reportsĀ “none of the phones are working” when in fact only 1 or 2 are not working

2. You can actually know when there is a problem with the system – where you otherwise might not know there is a problem until someone calls on your mobile to say your office number is not working

 

I have 2 scripts running every 15 minutes to email me with the details of any down extensions and trunks. This is done in Crontab with the line:

*/15 8-18 * * Mon-Fri /usr/Peermonitor.sh

and similar for Trunkmonitor.sh. This line says to run every 15 minutes between 08:00 and 18:00 every Monday-Friday

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DayNight aka Call Flow Control – More than 10

This information is provided without warranty – although I have been using this configuration successfully for over 12 months.

In FreePBX there is a module which has changed it’s name but remains an extremely useful one. Day/Night control, now called Call Flow Control, allows you to set a toggle-switch to change how a call is routed within the system. Typically this could be a Day or Night mode service, but you might also want a ‘We are closed for Christmas’ message for example. Using an announcement as the ‘night’ destination, using a recording linked to a feature code before going to a voicemail box, gives you a very quick way to temporarily close the office, with whatever message your users want to record themselves.

However, the module has a limitation; it only allows you to setup Callflows 0-9, for a total of 10 call-flow options. Should you need more than this, you would either need to program them manually, or else you can edit the module thus:

(This information was taken from here: http://www.freepbx.org/forum/freepbx/users/number-of-day-night-mode-controls-available-in-freepbx although this guide is a more basic version of the proposed patch)

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Using Aastra phones with Trixbox and Endpoint Manager

Below is a guide for getting Aastra phones up and running with a SysAdminMan Trixbox VPS using Endpoint Manager (You can always configure the phones manually instead of using Endpoint Manager). This guide uses 2 Aastra 55i’s and Trixbox 2.8 CE. This guide could also be used for configuring Endpoint Manager on other distributions.

Firstly you will need to allow access to the TFTP configuration files from your public IP address. You can see how to do this here – http://sysadminman.net/blog/2011/allowing-tftp-access-on-a-sysadminman-trixbox-server-2013

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FreePBX vs A2Billing

When you start looking at control panels for Asterisk it can be difficult to decide what you should be using – FreePBX or A2Billing.

While they are both web GUIs for setting up Asterisk, they are used for different things and which one to choose depends on your needs.

Here is a brief description of both to help you decide –

FreePBX

  • Used for setting up extensions and trunks for inbound and outbound calls
  • Includes lots of features of a traditional PBX – voicemail, IVRs, ring groups, queues etc.
  • Includes Call Detail Records (CDR) that logs all calls, their destination and duration

A2Billing

  • Used for billing for calls
  • Can be used to charge for calling card, sip user or regular outbound calls
  • The heart of A2Billing are the rate cards that include the per minute cost for all destinations allowed to be called
  • Least cost routing with multiple rate cards with the cheapest route being chosen
  • Admin and customer interfaces

So FreePBX is used to setup Asterisk with the features of a ‘traditional’ PBX and A2Billing focuses on billing for different types of calls.

It is also possible to combine the two and use A2Billing to account for outbound calls for extensions setup within FreePBX.

A2Billing is more complicated to setup than FreePBX. While it’s possible to setup an extension and trunk in FreePBX and start making calls very quickly there is quite a learning curve with A2Billing. Managing rate cards which hold all destinations and their cost can be quite complex.

All VPS templates offered by SysAdminMan include FreePBX – SysAdminMan, Trixbox, Elastix and PBX-in-a-flash

The following two templates include A2Billing – SysAdminMan and Elastix

Blocking Asterisk hacking/scanning attempts with fail2ban

Warning – if you follow these instructions fail2ban will, by default, be protecting you against other scans such as ssh attempts. This means though that if you get your IP blocked you will not be able to connect to your server from that IP. Ensure that you whitelist your IP by following the instructions at the end of the post.

Over the past few weeks we have seen a big jump in the scanning of VOIP servers. All of these scans are brute force scanning attempts that first scan for valid extension numbers and then to brute force guess the extension password by repeatedly trying different passwords.

Unfortunately Asterisk doesn’t have anything built-in to prevent these types of scans but it is very good at logging these attempts in the Asterisk logs. This means we can use a free utility called fail2ban and the linux iptables firewall to block IP addresses that make repeated failed login attempts.

Fail2ban is already included in PBX-in-a-Flash but we can also use it with other Asterisk distributions.

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Poor audio or music on hold with Trixbox

If you are using Trixbox (or probably any Asterisk distribution) with SIP trunks and the Dahdi dummy driver and experiencing poor audio or music on hold then it may be worth changing the ‘internal_timing’ setting.

This can be set be editing the file – /etc/asterisk/asterisk.conf file and making sure that the following 2 lines are not commented out (they should not have a ; in front of them) –

[options]
internal_timing = yes

In some cases this can improve the audio quality dramatically.

FreePBX IVR slow to respond

If you’re using FreePBX or one of the distributions that use it such as Trixbox, Elastix, PBX-in-a-Flash and are having a problem with IVRs being slow to respond it it is worth checking that you do not have “Enable Direct Dial” enabled for the IVR.

This option allows a customer to dial an extension number rather than an IVR menu option but this means that FreePBX has to wait to see if an extension number is being dialled, which can introduce a delay.

If you don’t need callers to be able to dial extensions from an IVR then you can turn this option off.

FreePBX - disable Enable Direct Dial

Trixbox 2.6 passwords – Changing on a Sysadminman VPS

It’s definitely a good idea to change the default passwords you get with your Sysadminman hosted Asterisk server.

Here’s how to do it if you’re using Trixbox 2.6.

Changing the passwords should be pretty much the same as it would be on a regular Trixbox server.

First you need to SSH (a secure command line connection) to the server. If you use Windows on your desktop a good program for making SSH connections is Putty. It’s free and you can find it here – http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html

For changing the server root password –

Use Putty, or another SSH client, to log into the server as the user root

Run the passwd command –

# passwd
Changing password for user root.
New UNIX password:
Retype new UNIX password:
passwd: all authentication tokens updated successfully.

For changing the Trixbox ‘maint’ password –

Use Putty, or another SSH client, to log into the server as the user root

Run the passwd-maint command –

# passwd-maint
------------------------------------------------
Set password for Admin section of trixbox CE GUI
User: maint
------------------------------------------------

New password:
Re-type new password:
Updating password for user maint

For changing the Voicemail ‘admin’ password –

Use Putty, or another SSH client, to log into the server as the user root

(Unfortunately there is no command for changing this password and you need to manually edit a file)

nano /var/www/html/recordings/includes/main.conf.php

Now find the line that contains (line 73) –

$ARI_ADMIN_PASSWORD ="old-password";

and change $ARI_ADMIN_PASSWORD to equal your required password. Now save the file.

For changing the WebMeetMe passwords –

WebMeetMe passwords can be changed through the WebMeetMe interface