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 –
and then browse to /var/spool/asterisk/monitor where the recordings are stored –
from there you can either delete the “.wav” files directly or move them off your VPS and on to your PC
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 –
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) –
Usage : whitelist.sh IPADDRESS (eg - whitelist.sh 192.168.1.1)
The version of A2Billing included on the SysAdminMan VPS template has been upgraded to A2Billing v1.9.2
See here for more information – sysadminman.net/distro-sysadminman.html
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
* All configuration settings moved to database
and here are some screen shots of A2Billing 1.4 –
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.
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 –
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/
If you’re looking for a reasonably priced SIP handset to use with your Asterisk system then the Linksys SPA-941 could be a good choice. I ordered one from ebuyer and delivery was within a few days.
Here is a picture of mine –
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
Our Asterisk Only VPS template has been updated to run Asterisk 1.6 and CentOS 5.3.
This VPS comes with the Dahdi dummy driver compiled meaning you can run IAX2 trunks and Conferences if required.
For more details on all Sysadminman VOIP VPS plans click here
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.