SYSADMINMAN

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) -

In some cases this can improve the audio quality dramatically.

If you lose or forget your a2billing 1.4 root password then there is a simple procedure for resetting it by editing the MySQL database.

To do this you’ll need to get SSH access to your server. If you use Windows then Putty is a good, free application for this.

1. Log in to your server as root

2. Determine the database name, user and password for your A2Billing database
Continue reading ‘Reset A2Billing 1.4 root password’ »

If you have upgraded your version of Asterisk and find that your IAX2 extensions no longer work then the cause could be a change to the IAX protocol. This was made to resolve a security issue that could result in a denial of service attack.

You will see this error in the Asterisk log file if you are suffering from this issue -

If you use FreePBX then Asterisk can be made to function the same as before by adding the following 2 lines to /etc/asterisk/iax_general_custom.conf -

You will also need to do a -

or restart Asterisk for the changes to take effect.

More information about the reason for this change and the implications for disabling call token checking can be found here –
http://svn.digium.com/svn/asterisk/branches/1.6.0/doc/IAX2-security.pdf

It is now possible to run “Skype for Asterisk” (SFA) from Digium on a Sysadminman VPS.  This allows calls to/from Skype users from your Asterisk server.

SFA can be installed on any of the VPS templates including Trixbox, Elastix, PBX-in-a-Flash (PIAF) and Sysadminman. Please note though that installing it on PBX-in-a-Flash requires upgrading the version of Asterisk to what PIAF considers an ‘unsafe’ release.

SFA licenses are $66 per channel from Digium and can be purchased here. You will need to purchase the number of licenses you require before SFA can be installed.

If you are an existing Sysadminman customer and wish to install SFA please raise a support ticket asking for a dummy eth0 device to be created for your VPS. This is required to license SFA.

Sysadminman can install and configure SFA on your VPS for a charge of £20.

Digium have released pricing details for Skype for Asterisk – http://store.digium.com/productview.php?product_code=1SFA0001

While it’s great software at $66 per channel it could work out very expensive if you need multiple channels.

I have made available a demo VPS so you can have a look and play with FreePBX 2.5 and A2Billing 1.4.1

Please note the following -

• Do not enter any personal/account information in to this VPS – it is viewable by everyone!
• The VPS may be reset back to it’s default state at any time
• You may be asked to accept a security certificate when you first connect. This is because connections to the server are secure over https but using a locally signed certificate
• There are resource limits placed on the VPS so it may be slower than a regular purchased sysadminman VPS
• Please do not change the passwords / Please play nice!

Continue reading ‘Live demo FreePBX 2.5 and A2Billing 1.4 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 -

Here is a php script designed to download the callwithus.com rate file and create a rate card in a2billing from it. It’s designed to work with the latest version of A2Billing – version 1.4.1

Here is a link to the script – http://sysadminman.net/misc/cwu-rates.html

If you copy and paste the script in to a file on your A2Billing server called /tmp/cwu-rates.php you can then run it by doing -

Before running it you will need to set the correct dabase, user and password name for a2billing at the top of the script. Also, the default markup for your ’sell rate’ is 50% but you can alter this by modifying $markup.

If  everything went well you should then find a new rate card created in A2Billing. You will need to set a trunk for the rate card and also assign it to a call plan before it can be used. Obviously check that the rates it has created look good too!

Also, I would suggest that before you run the script you take a backup of you A2Billing database – just in case things go wrong.

This script can be easily modified to work with other providers that provide a rate file that you can download.

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 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.