Tag Archives: VPS

Managing files on your VPS – clearing up disk space

An Asterisk PBX usually requires very little disk space. One of the few exceptions is for voicemail and recordings storage. If a customers VPS runs out of disk space you can pretty much guarantee that the space is being used by recordings.

The easiest way to either delete or mass archive the recordings is via SCP. This works over the SSH port (22) and you use the root/ssh login details that were provided in the welcome e-mail.

If you’re a Windows user then there a nice, simple, free application called WinSCP – http://winscp.net/eng/index.php

Just enter your VPS IP address and the root username and password –

winscp login

and then browse to /var/spool/asterisk/monitor where the recordings are stored –

winscp browse

from there you can either delete the “.wav” files directly or move them off your VPS and on to your PC

SysAdminMan FusionPBX password change and IP whitelist

SysAdminMan FusionPBX passwords

SysAdminMan’s FusionPBX servers have 2 levels of security. The first is a master web server username/password. The second is the built FusionPBX username/password.

The second username/password can be changed in the FusionPBX web GUI by selecting Accounts / User Manager.

The first web server password can only be changed via the SSH command line.

To do this log in to your server as ‘root’ using something like Putty

Then run the following command –

# admin-passwd
New password:
Re-type new password:

SysAdminMan FusionPBX IP whitelist

Each server also runs fail2ban and denyhosts. These provide security against hackers but it’s also possible to accidentally block your own IP address. To whitelist your IP address you can run the following command via SSH (see above) –

# whitelist.sh

Usage : whitelist.sh IPADDRESS (eg - whitelist.sh 192.168.1.1)

A2Billing upgraded to 1.4 on Sysadminman VPS

The version of A2Billing installed on the Sysadminman VPS template has been upgraded to version 1.4.1. It also has Asterisk 1.6 and FreePBX 2.5 installed.

Click here for more details

A2Billing 1.4 includes new features such as –

* Agent/Reseller module
* Redesigned web GUI
* New ticket system
* Dashboard
* All configuration settings moved to database

and here are some screen shots of A2Billing 1.4 –

a2billing 1.4.1 login page

a2billing 1.4.1 home page

a2billing 1.4.1 system settings

Skype for Asterisk testing with FreePBX

THIS TEST SERVER HAS NOW BEEN TAKEN OFF-LINE

I downloaded the Skype For Asterisk beta today from Digium. I think tomorrow (7/8/09) is the last day to sign up for the beta but the license you receive is valid until 31/8/09.

So far I’ve just been testing inbound calls, that is calls from a Skype user in to an Asterisk system

Please, give it a go yourself – my Skype user ID is *** and the call goes to an IVR

It was pretty easy to install the software, there are detailed instructions that come with it.

If you use FreePBX and put the Skype calls through to the correct context you can create inbound routes based on the Skype user ID and route the calls as you would normally.

skype inbound route

Once the calls are fed into Asterisk they can be treated just as any other incoming call.

My test system routes the calls through to a FreePBX IVR with 4 options –

  • Press 1 for the Skype For Asterisk test conference
  • Press 2 for music on hold
  • Press 3 for echo test
  • Press 4 for the speaking clock

You need to make the dial pad visible in Skype so the you can select the options –

dialling asterisk from skype

The first option is a conference room and the Skype for Asterisk beta license allows up to 10 concurrent calls so if you’ve got some friends on Skype please give it a go and let me know in the comments below how it works!

The Asterisk server is running on a VPS based in the UK so the quality may vary depending where you are calling from.

The music-on-hold are MP3’s and came from here – http://www.onhold2go.co.uk/

Installing Digium g.729 codec for Asterisk on an OpenVZ VPS

Installing Digium’s g.729 codec for Asterisk on an OpenVZ VPS requires an Asterisk friendly VPS provider. This is because the installation routine relies on there being an ‘eth0’ device on the server. This is not normally the case with OpenVZ where the network device is called venet0.

An ‘eth0’ device can be created on the VPS by running the following command (this is done on the OpenVZ server) –
(see here for more information – http://wiki.openvz.org/Asterisk_G729)

vzctl set $VEID --netif_add eth0 --save

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Trixbox, Elastix and Asterisk videos

There are some great videos around to give you an idea about what you can do with Asterisk and FreePBX.

Here are a selection –

Kerry Garrison, the senior product manager of Trixbox gives a quick tour of the installation and setup of Trixbox 2.2. The first half of this video concentrates on installing Trixbox but if you have a Trixbox VPS the hard work is done for you. Trixbox is now on version 2.4

Trixbox features. A nice run through of some of the features in FreePBX/Trixbox.

A good (and pretty long!) explanation of what you can do with Asterisk. This doesn’t include any information about FreePBX, the web based GUI.