• Install Asterisk Dsm 6 Release

I’m trying to install Freepbx with Asterisk 14 into a docker container based on CentOS 6, running on a Synology RS812+ box with DSM 6 and a pre-existing mariadb. I’ve got DAHDI and all the other prerequisites compiling and installing. How to install Let’s Encrypt SSL Certificate on Xpenology; Install Xpenology DSM 6.1.x on Proxmox; Quicknick is BACK!! And in the make of releasing a new loader soon!! How to Create Ubuntu Live USB in Windows; Setup XPEnology DSM 6.1 on VMware Workstation 12.

Updated by LinodeContributed by Nick Rahl

Report an Issue View File Edit File

What is Asterisk?

Asterisk is an open source private branch exchange (PBX) server that uses Session Initiation Protocol (SIP) to route and manage telephone calls. Notable features include customer service queues, music on hold, conference calling, and call recording, among others.

This guide covers the steps necessary to provision a new CentOS 7 Linode as a dedicated Asterisk server for your home or office.

Before You Begin

1. Create a CentOS 7 Linode in your closest data center (barring Atlanta (us-southeast), which does not currently support SIP servers). A 2GB Linode is enough to handle 10-20 concurrent calls using a non-compressed codec, depending on the processing required on each channel.

2. Ensure you have followed the Getting Started and Securing Your Server guides to prepare your Linode. Do not complete the steps to set up a firewall.

3. Update your system:

4. Disable SELinux and reboot your Linode. If you have Lassie enabled, your Linode will be back up and running in a few minutes.

Configure firewalld

1. CentOS 7 enables firewalld’s public zone for the default interface (eth0). SSH and DHCPv6 services are also enabled by default. To verify your current firewalld zone:

   That should return:

   And:

2. Add the SIP services.

   Note

   All the following firewalld rules contain the --permanent flag to ensure the rules persist after a system reboot.

3. Depending on your needs, you may want to add other related ports:

   • MGCP - If you use media gateway control protocol in your configuration.

   • RTP - The media stream - you can change this in /etc/asterisk/rtp.conf.

   • If you plan to use FreePBX to manage Asterisk, add the following rule:

   • IAX - If you need IAX, add the following rule. IAX is “Inter-Asterisk Exchange” and was meant to allow multiple Asterisk servers to communicate with one another. Some VOIP trunking providers use this, but most use SIP. Unless your VOIP provider requires it or you are running multiple Asterisk servers, you probably won’t need IAX or IAX2.

4. Verify your new configuration with:

   You should see the services and ports you just added in addition to default SSH and DCHPv6 services:

Install PJPROJECT

PJPROJECT is Asterisk’s SIP channel driver. It should improve call clarity and performance over older drivers.

1. Install build dependencies:

2. As a non-root user, create a working directory for the build:

3. Change to that directory:

4. Use wget to download the PJSIP driver source code:

5. Extract it:

6. Change to the newly created directory:

7. Specify the compiling flags and options:

8. Ensure that all dependencies are in place:

9. If make dep completes successfully, then build the plugin. It should only take a few minutes.

10. Install the packages:

11. Ensure the libraries have been properly installed:

    You should see:

Install Asterisk

1. Return to your build directory:

2. Download the latest version of Asterisk 16:

3. Untar the file:

4. Switch to the new Asterisk directory, replacing 16.1.1 if needed:

Enable MP3 Support

1. To use MP3 files for Music on Hold, install Subversion:

2. Run the configuration script:

Configure and Build Asterisk

1. In your build directory for Asterisk, run the configure script to prepare the Asterisk source code for compiling:

2. Start the build process. After a short while, you should see a menu on screen allowing you to configure the features you want to build.

3. If you want to use the MP3 format with Music on Hold, you should select Add-Ons, then use the right arrow to move to the right-hand list. Navigate to format_mp3 and press Enter to select it.

4. Select additional core sound packages and Music on Hold packages in the left menu, and enable .wav format for your desired language (ie. use the EN package for English.).

5. Press F12 to save and exit.

6. Compile Asterisk. When finished, you should see a message which says Asterisk has successfully been built.

7. Install Asterisk:

8. Install sample configuration files:

9. Configure Asterisk to start itself automatically on boot:

Test Connection

You now have a working Asterisk phone server. Fire up Asterisk and make sure it runs.

1. Start Asterisk:

2. Connect to Asterisk:

   You should see an output similar to the following:

3. To see a list of possible commands:

4. To disconnect type:

   Once disconnected, Asterisk continues to run in the background.

Install Asterisk Dsm 6 Release

Next Steps

Now that you have an Asterisk server running on your Linode, it’s time to connect some phones, add extensions, and configure the various options that are available with Asterisk. For detailed instructions, check out the Asterisk Project’s guide to Configuring Asterisk.

Join our Community

This guide is published under a CC BY-ND 4.0 license.