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FreePBX V14: Credentials - ChanSIPIn this article we will explain how to configure a FreePBX V14 Credentials trunk with Telnyx.CWritten by Customer Success Updated over a year agoFreePBX is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server. FreePBX is licensed under the GNU General Public License (GPL), an open source license. FreePBX can be installed manually or as part of the pre-configured FreePBX Distro that includes the system OS, Asterisk, FreePBX GUI and assorted dependencies.Additional documentation and resoruces:Instructions for Configuring a FreePBX V15 IP TrunkIn this activity you will:Pre-requisitesVideo WalkthroughSetting up your Telnyx SIP portal account so you can make and receive calls:1. Install your FreePBX V14In this section, you'll go through the steps you need to follow to install FreePBX.2. Configuring basic settings for your FreePBXIn this step, you'll configure your FreePBX V15 and connect it to Telnyx. To begin, notice that the main FreePBX screen will offer you four options:3. Configuring SIP settings for your FreePBXAt this point you can now work on confirming network settings and configuring your SIP trunks and extensions.4. Configure Extensions for your Free PBXIn this section, you'll configure all your PJSIP extensions.For testing purposes, you can now use your SIP client to register with FreePBX using the username, password/secret and local IP address of your FreePBX.5. Configure a Trunk for your FreePBX6. Configure Outbound and Inbound Settings for your FreePBX7. Configure outbound routing8. Configure inbound routingThat's it, you've now completed the configuration of FreePBX V14 IP Trunk and can now make and receive calls by using Telnyx as your SIP provider!Additional ResourcesReview our getting started with guide to make sure your Telnyx Mission Control Portal account is setup correctly!Additionally, check out:Related ArticlesConfiguring a FreePBX V13 Credentials TrunkFreePBX V13: PJSIP CredentialsFreePBX V14: IP

Features that Power Your ServersLearn about the features that make up our VPS & Dedicated Servers. FreePBX FeaturesWhat You Can Expect with Your ServerAre you new to FreePBX? FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk, the world’s most popular open source telephony engine software.To learn more about FreePBX please visit: www.freepbx.orgBelow is a short listing of the features that FreePBX has to offer:Unlimited ExtensionsYou can create an unlimited number of extensions via the FreePBX control panel.Completely SIP Based Unlimited SIP trunk capabilities are included.Paging and Intercom Support for group paging through your telephone handsets or integration through your overhead paging system.Company DirectoryEmpowers your customers to find people in your organization easily and connect directly to their extension.Unlimited IVR’s Auto Attendants allow you to direct your customers to various parts of your organizations through easy to use push button menus.Caller-ID See who is calling directly on your phoneSoft-Phone Support You can use software on your personal computer or smart phone and receive/make calls through your PBX.Voicemail to email Receive you voicemail messages in your email box and play them over your computer speakers.Music On Hold Play music on hold to callers. Optionally use different classes of music on hold depending on which Queue a caller is in.Automatic Backup Backup your PBX automatically every night. If something goes wrong we can have you up in running from a backup in a matter of hours.Voicemail Config Simply log into your voicemail and walk through recording your greetings.Call Flow Control Call Flow Control makes it easy to take control of your business hours. Need to open early or stay late? Simply press a button to keep calls flowing in. Need to leave early? Press the Call Flow Control Toggle on your phone and route all calls to your closed IVR or voicemail.FreePBX Special Offer30 Day Money Back GuaranteeWe offer a 30-Day Money Back Guarantee. If you are not 100% satisfied with our service we will refund your money if it is within 30 days of your initial account setup. In the event that the Customer terminates service within 30 days under this trial, all fees shall be waived for Customer; and furthermore, Customer shall receive a credit for any fees, if any, invoiced and/or paid at the time of the termination of this Service. Should Customer not exercise the right to terminate service within 30 days under this trial, all fees indicated on this Agreement shall be due and payable.Find Your FreePBX Hosting PlanBrowse our plans to find the right phone system for your business.Sangoma and CyberLynk SupportWorld Class Support for Your FreePBX ServerTogether CyberLynk and Sangoma provide world class support for every customer. CyberLynk handles the datacenter operations, server hardware, billing and issues directly related to your VPS or dedicated server. Sangoma provides direct support for all of your FreePBX questions.Mobile AccessVPS Control Panel Access Via SmartphoneOn the road and wish you could reboot your FreePBX VPS? Problem solved. You can access the mobile version

Flexible compared to pure open-source components.Ideal Use Cases Elastix hits the sweet spot for:Small businesses that want a full-featured phone system.Companies that need unified communications beyond just a PBX.Organizations without dedicated IT resources to manage separate tools.For these customers, Elastix delivers a VoIP phone system with built-in conferencing, fax, and messaging – all neatly wrapped up and easy to manage through the web GUI.4. FreePBXFreePBX is an open-source GUI interface that manages and configures the powerful Asterisk PBX. It provides an easy-to-use administrative interface for setting up and controlling your phone system without needing deep technical skills.Think of FreePBX as a front-end for accessing the advanced features already built into Asterisk. You still need to have Asterisk installed, but FreePBX makes it much more accessible to administrators through the web.FeaturesSimplified interface for initial setup of extensions, trunks, routes, etc.Easy-to-use control panel for administrators to make changes on the fly.Configuration wizards for setting up call routing, IVR, voicemail boxes, and more.Seamless integration with popular SIP telephony devices.Add-on manager for installing plugins and commercial modules.Multi-language support for global deployments.Pricing & SupportThe core FreePBX platform is 100% open source and free. Paid commercial modules are available from Sangoma to add more enterprise-level features. Paid professional support subscriptions are also available if desired.ProsA streamlined way to set up and manage Asterisk.Great for users less familiar with Linux administration.Active development community behind the project.Commercial modules are available if more features are needed.ConsStill requires setup and management of the underlying Asterisk instance.Paid modules can add up in cost for large deployments.Less flexibility compared to editing Asterisk configs directly.Ideal Use Cases FreePBX is a great choice for:SMBs that want an easy-to-use phone system interface.Companies with less technical resources to devote to telephony.Organizations that prefer a commercial support option.Asterisk power users who still want a control panel for quick changes.If you need the advanced capabilities of Asterisk but don’t have an army of Linux gurus to manage it full-time, FreePBX can provide the usability layer on top.5. YATEYATE (Yet Another Telephony Engine) is a modular open-source VoIP PBX built using the Enhanced Nethack Telephony Interface (ENTI) framework. It uses a lightweight stand-alone architecture optimized for speed and efficiency rather than being a fully featured PBX system.The modular design allows features to be flexibly added or removed as needed for custom solutions. YATE has been ported to run on Linux, BSD, and other UNIX-based operating systems.FeaturesBuilt-in SIP registrar and proxy server for VoIP calls.IVR module for interactive voice response menus.Voicemail and call recording modules available.Real-time billing and routing capabilities.Can send and receive SMS messages.APIs for building custom telephony applications.Pricing & Support YATE is available for free under the GPL open-source license. There is community-based support available through forums and the project wiki. Paid support plans are offered by some commercial partners and consultants.ProsVery lightweight and efficient compared to large PBXs.Flexible for creating custom telephony solutions.Can connect to standard SIP desktop and mobile clients.Runs well on low-resource machines like Raspberry Pi.ConsNot a full PBX replacement for large enterprises.Configuration requires editing