This setup is often referred to as a SIP trunk. You will buy a phone number (often called a direct inward dial, or DID), and you’ll configure your phone system to connect with the provider’s phone system via SIP. No matter who you choose, the process is fairly straightforward. How well do you know Linux? Take a quiz and get a badge.Linux system administration skills assessment.A guide to installing applications on Linux.Download RHEL 9 at no charge through the Red Hat Developer program.I will be using VoIP.ms for the phone numbers in this article. Larger organizations may choose to use a local telecommunications provider or their local Internet company, but there are also a variety of online providers available (e.g., Twilio and VoIP.ms, among many others). Obtaining a telephone number and connecting your phone system to the PSTN might sound like a difficult task, but it’s actually pretty easy. Be sure to read the documentation that comes with the Asterisk source code and understand every piece of configuration that you add to Asterisk. Connecting to a telephony provider can open you up to all sorts of issues, such as toll fraud, if you aren’t careful about controlling who can connect to your phone system (e.g., via firewall rules). Note: I’ve mentioned it several times throughout this series, but it is especially important for this article to understand how to secure your phone system. In this article, you will learn some background about how to connect to the PSTN and you will see the Asterisk configuration needed for this connectivity. In two previous articles, you learned how to configure two SIP phones and the Asterisk dialplan to enable the phones to call each other. Having two phones that can call each other is great, but most organizations want to connect their phone system to the public switched telephone network (PSTN) to allow for inbound and outbound calling to others outside of the organization.
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