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| <!DOCTYPE html> | ||
| <html lang="en"> | ||
| <head> | ||
| <meta charset="UTF-8"> | ||
| <meta name="viewport" content="width=device-width, initial-scale=1.0"> | ||
| <title>Hands-on Lab - VoIP Phone and SIP Configuration</title> | ||
| <link rel="stylesheet" href="https://github.com/skunkworksza/Courses/blob/01882983f090495a5d0f3549bd04c378aa3a3398/css/main.css"> | ||
| <link rel="stylesheet" href="https://unpkg.com/carbon-components/css/carbon-components.min.css"> | ||
| <style> | ||
| body { | ||
| font-family: 'IBM Plex Sans', sans-serif; | ||
| background-color: #f4f4f4; | ||
| color: #333; | ||
| margin: 0; | ||
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| color: white; | ||
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| box-shadow: 0 4px 8px rgba(0, 0, 0, 0.1); | ||
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| footer { | ||
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| </style> | ||
| </head> | ||
| <body> | ||
| <header> | ||
| <nav> | ||
| <ul> | ||
| <li><a href="/">Home</a></li> | ||
| <li><a href="/training">Training</a></li> | ||
| <li><a href="/services">Services</a></li> | ||
| <li><a href="/contact">Contact</a></li> | ||
| </ul> | ||
| </nav> | ||
| <h1>Hands-on Lab - VoIP Phone and SIP Configuration</h1> | ||
| </header> | ||
| <main> | ||
| <section> | ||
| <h2>Introduction</h2> | ||
| <p>In this hands-on lab, you will configure a VoIP phone and set up SIP (Session Initiation Protocol) in Asterisk. By the end of this lab, you will be able to:</p> | ||
| <ul> | ||
| <li>Configure and register a VoIP phone with Asterisk.</li> | ||
| <li>Set up SIP settings in Asterisk.</li> | ||
| <li>Test and troubleshoot VoIP phone and SIP configurations.</li> | ||
| </ul> | ||
| <p>This practical exercise will provide you with the skills needed to deploy and manage VoIP phones and SIP configurations using Asterisk.</p> | ||
| </section> | ||
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| <section> | ||
| <h2>Step-by-Step Instructions</h2> | ||
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| <h3>1. Configure VoIP Phone</h3> | ||
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| <h4>Access VoIP Phone Web Interface</h4> | ||
| <p>Connect your VoIP phone to the network. Open a web browser and enter the IP address of the VoIP phone (e.g., <code>http://192.168.1.100</code>).</p> | ||
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| <h4>Log In to the VoIP Phone Interface</h4> | ||
| <p>Enter the default username and password (check the phone's manual for these details).</p> | ||
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| <h4>Configure SIP Account on VoIP Phone</h4> | ||
| <p>Navigate to the SIP settings or Account settings. Enter the following details:</p> | ||
| <ul> | ||
| <li>SIP Server: <code><IP_of_Asterisk_Server></code></li> | ||
| <li>Username: <code>1001</code></li> | ||
| <li>Password: <code>welcome123</code></li> | ||
| </ul> | ||
| <p>Save the settings and apply the changes.</p> | ||
| </section> | ||
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| <section> | ||
| <h3>2. Test VoIP Phone</h3> | ||
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| <h4>Make a Call Using the VoIP Phone</h4> | ||
| <p>Dial <code>1002</code> from the VoIP phone and press <code>Call</code>. Verify that the call is received on the SIP client configured for 1002.</p> | ||
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| <h4>Check VoIP Phone Registration</h4> | ||
| <p>Open Asterisk CLI:</p> | ||
| <pre><code>sudo asterisk -r</code></pre> | ||
| <p>Check the SIP registration status:</p> | ||
| <pre><code>sip show peers</code></pre> | ||
| </section> | ||
|
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| <section> | ||
| <h3>3. Troubleshoot SIP Configuration</h3> | ||
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| <h4>Verify SIP Settings</h4> | ||
| <p>Ensure the SIP server address, username, and password are correctly configured on the VoIP phone.</p> | ||
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| <h4>Check Network Connectivity</h4> | ||
| <p>Ensure the VoIP phone is properly connected to the network and can reach the Asterisk server.</p> | ||
| </section> | ||
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| <section> | ||
| <h2>Conclusion</h2> | ||
| <p>By following these detailed hands-on labs, you will build a comprehensive understanding of Asterisk, starting from basic installation and configuration to advanced telecommunication solutions. Each lab builds upon the previous one, ensuring a smooth learning curve and avoiding redundancy.</p> | ||
| </section> | ||
| </main> | ||
| <footer> | ||
| <p>© 2024 Skunkworks. All rights reserved.</p> | ||
| </footer> | ||
| <script src="https://unpkg.com/carbon-components/scripts/carbon-components.min.js"></script> | ||
| </body> | ||
| </html> |