Guide to install and configure Asterisk in CentOS 6.6

To begin I will give a brief description of what it is Asterisk in case someone doesn't know.

• Asterisk is a free software created by Mark Spencer founder of the company Digium the main developer, which performs functions of PBX. Una PBX in short, it is a telephone exchange.
• Asterisk allows you to manage calls, mailboxes, create a IVR (Interactive voice response) between many other things.
• It recognizes a lot of protocols VoIP (Voice over IP) among which are SIP e IAX which are the most important.

Now that we know about Asterisk let's install and configure it for a CentOS 6.6 server

The installation part is different depending on the distro we use but the configuration is the same for Ubuntu, CentOS, Debian, etc.

Installation

The first thing we are going to do is add the repository inside /etc/yum.repos.d/ :

• The first repository that we are going to introduce will be: centos-asterisk.repo

[asterisk-tested] name=CentOS-$releasever - Asterisk - Tested
baseurl=http://packages.asterisk.org/centos/$releasever/tested/$basearch/
enabled=0
gpgcheck=0
#gpgkey=http://packages.asterisk.org/RPM-GPG-KEY-Digium

[asterisk-current] name=CentOS-$releasever - Asterisk - Current
baseurl=http://packages.asterisk.org/centos/$releasever/current/$basearch/
enabled=1
gpgcheck=0
#gpgkey=http://packages.asterisk.org/RPM-GPG-KEY-Digium

• The second repository will be: centos-digium.repo

[digium-tested] name=CentOS-$releasever - Digium - Tested
baseurl=http://packages.digium.com/centos/$releasever/tested/$basearch/
enabled=0
gpgcheck=0
#gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium

[digium-current] name=CentOS-$releasever - Digium - Current
baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/
enabled=1
gpgcheck=0
#gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium

Once this is finished we do a
yum update

Now we will install the package asterisknow-version.noarch

yum install asterisknow-version

Installing this package we add the rest of the Asterisk repositories, with all their available versions. Now we will see that new files have been created in the /etc/yum.repos.d folder. We are going to install the version 13 de Asterisk.

To choose a specific version (by default it uses the repository of version 11) we use the parameter –enablerepo (it enables the repository that we indicate) in the installation with yum, that is, we can install any version indicating the appropriate repository
yum install asterisk asterisk-configs asterisk-sounds-core-es-gsm --enablerepo=asterisk-13

How are we going to install the latest version of Asterisk, we will enable the default repository to update Asterisk to be centos-asterisk-13.repo.

To do this we modify the enabled parameter of the repository centos-asterisk-11.repo a 0, so that it is disabled. And in the same way, we modify the enabled parameter of the repository centos-asterisk-13.repo a 1, so it will become the repository used for updates to Asterisk. Depending on the version that we have installed, we will enable the repository of our version.

Configuration

We go to the configuration folder that is in / etc / asterisk

cd /etc/asterisk/

We will find many files of which right now we will use three:

• sip.conf → In it the SIP users that we are going to use are registered
• extensions.conf → In it we register the extensions
• voicemail.conf → In it we register the mailboxes

We start with the file sip.conf

We go down to the end of the file or wherever we want, we choose the end for the convenience of knowing where they always are. And we continue the following structure for each user:

[ejemplo] type=friend
secret=pass
qualify=yes
nat=no
host=dynamic
canreinvite=no
context= contexto
mailbox= buzon


• We must put the User nameIn this case example
• La password of the user is put in secret
• En host we have decided to put dynamic so that the user can connect from any IP on the network
• Context is the context, we put the context in which it will be
• Mailbox it's voicemail

Now I am going to leave you an example of what I have done
[rack] type=friend
secret=1234
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=iesjg
mailbox=00@buzon

Over the line 343 of the sip.conf we establish language = isWith this we make all the users that we believe have Spanish as their default language.

Now we are going to configure the voicemail.conf

To configure the mailboxes we go down to the bottom and create a context, for example [mailbox] and thus configure the mailboxes with the following structure:
número => pass,nombre,correo

And now the example that I have done:

00 => 1234,rack,rack@iesjorgeguillen.es

Already for last let's configure the extensions.conf

At the bottom of the document, in my case, I will add the configuration

We will create a context in my case [iesjg], remember that the contexts are in brackets.

And now within that context we will create extensions such as:
exten => 00,1,Dial(SIP/rack,15,Ttm)
same => n,VoiceMail(00@buzon)
same => n,Hangup()

• What it does is every time they call extension 00 first it calls the SIP / rack (which is created in sip.conf) for 15 seconds with background music and the call can be forwarded to another phone
• We use "Same" to avoid having to be putting "exten => 00"
• If not answered, in the next place "n”Skip the mail with the function VoiceMail (00 @ mailbox) previously set to voicemail.conf
• And finally we do a Hangup () to hang

Now I am going to put some useful Asterisk functions.

1. Dial()

• With this function, make a call to where we pass in this case SIP / rack

2. Voicemail()

• With this function the voice mailbox of that user is called to leave a message (mailbox number@contexto_en_voicemail.conf)

3. hang-up()

• With this function we hang up.

4. Wait()

• With this function a wait is carried out, the time we want to pass Wait (1), wait 1 second.

5. VoiceMailMain()

• Call to menú of the voicemail, it will ask you the extension and its password to listen to the messages.

6. Answer()

• With this function the Asterisk PBX answers the call.

7. All time lap record()

• With this function an audio is recorded. For example to create a menu.

8. background()

• Play an audio in the background.

Now we restart the service.
service asterisk restart

And now we run the asterisk console to carry out monitoring.

asterisk -rvvvvvvvvvvv

When we connect phones with the created SIP users we can see them in the console by executing:

sip show peers