As someone who is eager to experiment with VOIP, I'm currently lacking information on how to actually set up a server for testing purposes. I have many questions about the process, such as which phone is appropriate for calls and how to terminate calls to an external phone. While I am familiar with terms like Asterix, FreePBX, Trixbox, and A2billing, I've found that most online resources are geared towards those who already have a solid understanding of VOIP technology.

If you are a VOIP expert, I would love any assistance you could offer in pointing me towards useful resources for newcomers such as myself. Whether it be articles, books, publications, or anything else you think might be helpful, I am open to any suggestions you may have.

For context, I am a Linux server administrator and feel comfortable working with this type of technology.

Although VOIP terminology can be overwhelming, the good news is that starting with a simple setup is fairly easy. To begin calling and creating from there, I recommend getting a SIP client, as most phones and carriers use this protocol for communication. Fortunately, there are many SIP softphone clients available for download, such as Linphone, which works on most platforms.

After acquiring a SIP client, you will need an ITSP (internet telephone service provider) to act as your gateway to the regular telephone network (PSTN). There are numerous ITSPs to choose from, such as voip.ms, where you can top up your account with credit and pay for calls as you go. Purchasing a dialer number (DDI/DID) for a small monthly fee will enable incoming calls to your SIP clients.

Once you have gained some experience, you may want to explore more advanced options like Asterisk/FreePBX. These PBX systems sit between your internal line and telecom operator and offer features like voice messages, call recording, search groups, and free extensions on extension.

Firstly, since you are comfortable working with Linux, there are several open-source VOIP software options that you can explore. One popular choice is Asterisk, which is a free and powerful open-source telephony platform. Asterisk allows you to build your own VOIP server and offers a wide range of features and functionality.

To set up a VOIP server using Asterisk, you can start by installing the necessary packages on your Linux server. The official Asterisk website provides documentation, tutorials, and sample configurations to help you get started. They offer resources specifically for beginners, so that might be a good starting point for you.

Another option is FreePBX, which is a web-based graphical user interface (GUI) that sits on top of Asterisk. It simplifies the process of managing and configuring Asterisk, making it more beginner-friendly. FreePBX has its own documentation and community forums where you can find help and resources.

Trixbox and A2Billing are also worth exploring as they are based on Asterisk and provide additional features and functionality. However, keep in mind that Trixbox is no longer actively maintained, so you may want to consider other options.

In addition to the official documentation for these platforms, there are several online communities where you can find helpful information, tutorials, and troubleshooting tips. Websites like voip-info.org, forums such as the Asterisk community forum, and subreddits like r/VOIP can be great places to gather information and ask questions.

Lastly, if you prefer learning from books, there are several resources available on Amazon or other online bookstores. Some recommended titles include "Asterisk: The Definitive Guide" by Russell Bryant, Leif Madsen, and Jim Van Meggelen, and "FreePBX 101: A Beginner's Guide to Free and Open Source VoIP PBX" by Darren Schreiber.


few more tips and suggestions for setting up a VOIP server:

1. Hardware Requirements: Make sure your server hardware meets the basic requirements for running Asterisk or any other VOIP software you choose. It's important to have a suitable CPU, sufficient memory, and reliable storage.

2. Network Considerations: VOIP is highly dependent on network quality and bandwidth. Ensure that your server has a stable internet connection with adequate upload and download speeds. If possible, consider using Quality of Service (QoS) settings to prioritize VOIP traffic on your network.

3. SIP Trunk Providers: To terminate calls to external phones, you may need to sign up with a SIP trunk provider. These providers offer connectivity services that allow your VOIP server to communicate with traditional phone networks. Some popular SIP trunk providers include Twilio, Flowroute, and Telnyx.

4. Phone Selection: When it comes to choosing the appropriate phone for making VOIP calls, there are numerous options available. You can use softphones (software-based phones), hardware-based IP phones, or even smartphone apps. Popular softphone options include Zoiper, X-Lite, and Linphone. For physical IP phones, brands like Yealink, Grandstream, and Cisco are widely used.

5. Security Considerations: VOIP servers can be vulnerable to attacks, so it's important to implement security measures. Configure strong passwords, keep your software up-to-date, and consider using a firewall to protect your server. Regularly monitor logs and take necessary actions to mitigate security risks.

6. Virtualization: If you prefer to experiment with VOIP in a virtualized environment, you can use platforms like VMware, VirtualBox, or Proxmox to set up virtual machines for your server. This allows you to easily manage and test different configurations.