Elastix sip trunk

Setting Up Elastix SIP Trunk: A Comprehensive Guide

Elastix is a robust open-source software designed to integrate the best tools for IP telephony in one interface, making it a popular choice for businesses looking to establish reliable communication systems. One of the essential features of Elastix is its ability to support SIP trunks, which are crucial for connecting an IP PBX to the traditional PSTN or to other VoIP networks. Here’s a step-by-step guide to setting up a SIP trunk in Elastix.

What is a SIP Trunk?

A SIP trunk is a virtual phone line that uses the Session Initiation Protocol (SIP) to enable voice over IP (VoIP) communications. It connects your PBX to the Internet and allows for making and receiving calls globally without traditional phone lines.

Prerequisites

1. Elastix Installation: Ensure you have Elastix installed and properly configured on your server.
2. SIP Trunk Provider: Choose a SIP trunk provider that fits your needs. Obtain the necessary account information, including the SIP server address, username, and password.

Setting Up the SIP Trunk in Elastix

1. Log into Elastix:
   • Access the Elastix web interface through your browser.
   • Enter your credentials to log in.
2. Navigate to PBX Configuration:
   • Go to the “PBX” tab in the main menu.
   • Select “PBX Configuration.”
3. Add a SIP Trunk:
   • Click on “Trunks” in the left-hand menu.
   • Click on “Add SIP Trunk.”
4. Configure SIP Trunk Settings:
   • General Settings:
     • Trunk Name: Enter a name for your SIP trunk (e.g., “MySIPTrunk”).
   • Outgoing Settings:
     • Trunk Name: Enter the same name as above.
     • PEER Details: Input the details provided by your SIP trunk provider. Typically, this includes parameters like:
       makefile

       Copy code

       host=sip.provider.com
username=your_username
secret=your_password
type=peer
qualify=yes
nat=yes

   • Incoming Settings:
     • Often, these can be left blank, but if your provider has specific instructions, follow them.
   • Registration String:
     • Provided by your SIP trunk provider, typically in the format:
       graphql

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       username:password@sip.provider.com

5. Save and Apply Changes:
   • Click “Submit Changes.”
   • Go to the “Apply Configuration Changes Here” link at the top Chinese Overseas Asia Number to reload the configuration.
6. Configure Outbound Routes:
   • Go to “Outbound Routes” in the PBX Configuration menu.
   • Add a new route and define the dial patterns that will use the SIP trunk.
   • Select the SIP trunk you just created in the “Trunk Sequence” for Matched Routes.

Testing the SIP Trunk

1. Make a Test Call:
   • Use a connected IP phone or softphone to make an outbound call.
   • Ensure the call routes correctly through your SIP trunk.
2. Check Call Logs:
   • Navigate to “Reports” and then “CDR Reports” to verify that Armenia Phone Number the call was logged and routed through the correct trunk.

Troubleshooting Tips

• Check Firewall Settings: Ensure that the necessary ports for SIP (typically 5060) and RTP (media) are open.
• Verify Credentials: Double-check the username, password, and server address for any typos or errors.
• Consult Provider Documentation: Refer to your SIP provider’s setup guide for any provider-specific settings or configurations.

By following these steps, you can efficiently set up a SIP trunk in Elastix, enhancing your business communication capabilities with a flexible, cost-effective solution.