If you have implemented Music on Hold on your asterisk box, you may want to listen to it, to know if it is working as you want, of course you can call to your IVR number and wait for the music on hold, but it is better if you create an extension specially designed to hear to music on hold.

This is how to do it, in the extensions.conf file add these lines in the context your extension is.

The most common way to use an asterisk VoIP PBX is with an internal FXO card like Digium or Sangoma, but there are other ways, like using an external gateway like Quintum or Audiocodes to name just two of the options.

I am not an Asterisk expert, but I prefer the second aproach, as it may save CPU power on the Asterisk server, as the Analog to Digital and Digital to Analog conversion is now done in the external gateway instead of the Asterisk server.

Having said that, I will show you how to route calls to that external gateway, keep in mind that may be other ways to do this, and maybe better ways, but this is the one that worked for me.

I have been working these last days on an Asterisk over Debian, and I will be posting some tips I am learning about it, today I will post about how to record a voice message that can be used with the IVR menu.

The IVR (Interactive voice responce) is one of the most used PBX applications, and the best VoIP PBX (Asterisk) have a real good support of this feature.

Continuing with the Asterisk setup, today we will see how to configure Ekiga to work with it, to install Ekiga, launch your package manager and use it to install Ekiga.

This time I will use a screencast to show you this how-to, this is my first try, so please be gentle if you see that is in poor quality, or if it could have been better edited, I am sure the next ones will start to improve, hope you may find this how-to useful.

Please be sure to read: simple Asterisk configuration to understand the parameters used in this video.

Once you have asterisk installed and running you need to configure it, to be able to use it as PBX.

Asterisk has a lot of features, and we will start to explore some of them here step by step, in this first post about configuring Asterisk, I will only show you how to configure a single two internals and make them be able to call each other, you may add more than one, as you will see later, and all of them will be able to call each others.

So lets start:

In the past I have shown you, how to install Asterisk on Debian and CentOS, Now I will show you how to do it on Arch Linux

According to wikipedia:

Asterisk is a software implementation of a telephone private branch exchange (PBX) originally created in 1999 by Mark Spencer of Digium. Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services. Its name comes from the asterisk symbol, "*".

Asterisk also supports a wide range of Video[2] and Voice over IP protocols, including SIP, MGCP and H.323. Asterisk can interoperate with most SIP telephones, acting both as registrar and as a gateway between IP phones and the PSTN

Well, to install it on Arch Linux, we will use the AUR Tarball, as there is still no binary available to install using the normal pacman repositories.

I am starting to work with Asterisk which is a Linux VoIP PBX, that lets you even send your minutes traffic, to VoIP gateways, and therefore one more time Linux is helping us lower the costs of maintaining an office.

As a first thing I tried to install it from repositories both on Debian and CentOS.

Let's start with Debian.