freepbx end user manual

For conferences that need more than three participants, conference rooms should be used (see above). This is called local conferencing and does not use a teleconference room. This softkey allows a quick way to have a conference call with up to two other people. All parties involved will have the call end. For example, the system may be configured to direct calls during office hours directly to the receptionist and to route calls to voice menu after hours. This override is usually used if the receptionist needs to be away unexpectedly or if the office needs to close due to poor weather. The caller can then leave a voice message. It is an alternative to setting up call forwarding (above). The difference is call forwarding will immediately redirect the call without ringing your normal extension, while Follow me will ring both your extension and your alternate numbers, so you can answer the call either way. Paging mode is a one-way voice announcement to all (or selected) phones. Intercom mode allows for a two-way conversation between two or more phones. By default, these features will not interrupt normal calls currently in progress. Parking calls This feature allows an incoming call to be parked and then picked up by any other extension. This is an alternative to transferring a call directly to a particular extensions. Some organizations prefer to park the incoming call and make a general page asking the desired person to pick up the call from wherever they are in the building using whatever phone is closest to them. It needs to be enabled on the phone system before use (please contact your phone system administrator for this). Note that certain legal jurisdictions require the other party on the call to be informed the call is being recorded. Access the recorded audio Records are accessed through the user portal web site (this is the same site that also provides web access to voicemail settings).

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If available, using dedicated phone handset keys is the recommended approach, as this approach will be easier to use and easier to remember. Handling calls Here are instructions on how to handle incoming and outgoing calls. The caller on line 1 will automatically be placed on hold. If the handset is lifted at the time, the call will be placed in handset mode. If the handset is down, the call will be placed in speakerphone mode. Call forwarding This feature allows you to direct calls to your extension to instead to another internal extension or to an external phone number, such as your cell phone or home phone number. Two types of call transfers are available: blind transfers and attended transfers. Blind Transfers Blind transfers are where the call is transferred without you needing to stay on the line. Answer the call The caller indicates they would to be transferred to someone else Press the dedicated transfer softkey for the desired destination extension The caller will immediately be directed to the new extension, which will begin to ring Your connection will automatically end. This is sometimes also called a consultative transfer. The outside caller will remain on hold. Just hang up. Aastra 6737i dedicated softkey instructions If dedicated transfer softkeys for various extensions has been set up by your phone system administrator, the softkey can be used to perform attended transfers without having to enter the internal destination extension number. The system can be set up for as many conference session rooms as are needed. As many people can join a conference room session as you have internal and external lines available. Please contact your phone system administrator for additional configuration options. Conferencing (using your phone only) The Aastra phones include the built-in capability to set up an ad-hoc conference call with a maximum of three parties (the original caller, yourself, and one additional party).

A single user can only be linked to one extension, and one extension can only be linked to a single user. User Details All of the fields in this tab are optional. This information is stored for use in other apps. Most fields are self-explanatory. The Display Name is a customized username that will be displayed in UCP and Contact Manager. You will need to enter an Email Address for the user if you plan to have the system send the user a welcome e-mail message. FreePBX Administration GUI Here you can give a user administrative access to your PBX. You can control the user's level of privileges. Visible Extension Range (Optional) Enter a starting number and ending number if you want to restrict this user's view to only the extensions, ring groups, and queues within this range. You can only enter one range. Administration Access Select the Admin Sections this user should have access to. The default is None selected. If you need to grant access to modules, click the None selected button to bring up a drop-down menu. Then either click the check box next to Select all (for all modules), or click the check box(es) next to the module(s) this user will be allowed to access. The Fax tab is only available if the user has been linked to an extension. If the user hasn't yet been linked to an extension, select the primary linked extension in the Login Details tab, submit settings and apply config, and you should now see the Fax tab. Attachment Format The format to convert the incoming fax file to before e-mailing it. Outgoing Station ID The outgoing Fax Machine Identifier. This is usually your fax number. Outgoing Header The outgoing Fax Machine Header. This usually contains the name of the person or entity sending the fax. Coversheet name The name that will appear on the outbound cover sheet. Coversheet Tel The phone number that will appear on the cover sheet. Coversheet Email The email address that will appear on the coversheet.

It's helpful if you have recorded the pronunciation of your name in your voicemail settings (please see Voicemail for instructions) so callers can hear your name spoken normally instead of hearing the phone system spell out your name letter-by-letter. Voice mail Please see the Voicemail section for a full description of how to use the voice mail system. Other operations There are a number of other less-used features not specifically listed above. The feature code reference list shows all the available phone system feature codes. Please contact your phone system administrator if any of these features are desired. See Feature Codes Module for more information. View the current version. In FreePBX Version 12 and newer, the new User Control Panel (UCP) utilizes the User Management module. UCP replaces the old ARI in FreePBX. For example, a user could be allowed to log into the User Control Panel and access the voicemail of three other accounts or manage your API tokens. Note In FreePBX 13, fax options have moved from the Extensions module to the User Management module. Logging In On the top menu click Admin In the drop down click User Management The Users tab should be active, and a list of current users should be displayed. To edit an existing user, click the edit button. You can also create a new user manually. In the Users tab, click the Add button. This should take you to the Login Details tab for the new user. Description A friendly name or brief description for this user. Password Password for the user. Note: If you proceed to enable XMPP later, the password will be exposed in the database, and you will need to redefine it. Groups A list of groups the user belongs to, if any. To select a group, begin typing the group name into the field, and when the system finds it, click on the name. You can add multiple groups. Primary Linked Extension This is the extension this user is linked to.

Enabling presence allows a user to select their presence state from a drop-down menu in UCP, which changes call behavior according to the selected state. You can customize the available presence states in the Presence States module. For more information on what this means, see our wiki What is Presence. SIPStation SMS SMS Assigned DIDs Select the DID(s), if any, that you would like to give this user access to for text messaging (SMS) within UCP. By default, none are selected. To add a DID, click the None selected button and choose one or more DIDs from the drop-down menu. Allowed Voicemail Enter the extension(s) whose voicemail this user will be able to access. WebRTC WebRTC, or Web Real-Time Communication, is a technology that enables browser-to-browser voice calls and other services such as video chat and P2P file sharing. In FreePBX, you can enable WebRTC to allow users to make web-based phone calls in UCP. In many cases this can cause issues. This is the certificate the phone will use for this user. XMPP This section controls whether the user has access to XMPP in UCP. XMPP stands for Extensible Messaging and Presence Protocol, which is a communications protocol for message-oriented milddleware XML. You will need to redefine the user's password after clicking Yes. Users The first view you will see when going to the User Management module is the listing of all of your users on the system You can see users for a specific directory by changing the directory drop down to another directory For more information on Users see User Management Users Groups Upon clicking the 'Groups' tab in the User Management module you will see the listing of all of your groups on the system You can see users for a specific directory by changing the directory drop down to another directory For more information on Groups see User Management Groups Directories A directory is a listing of Users and Groups.

Email Results When to email the results of an outgoing fax. Retries Number of times to retry sending a failed fax. Rest API This section is used in conjunction with the RESTful Phone Apps module, which gives you access to a range of phone-side applications. It requires the commercial End Point Manager (EPM) to configure your devices, as it needs to receive data from EPM in order for most applications to work correctly. If RestAPI is enabled here for a user, it will create a token and define the restrictions of what this REST API user has access to. If you enable RestAPI, you will gain access to the rest of the options below. Extensions Enter the extension(s) this user will be able to control. Click on the Extensions field to display a drop-down menu of available extensions. Click on an extension to select it. To add more extensions, repeat the process. Modules Select which modules the user will have access to. Use the checkboxes to select modules. Rate Limit Enter the quantity of API requests that this user can make per hour. The default is 1000. UCP User Control Panel (UCP) is a user interface for FreePBX. This section has several tabs that control a user's privileges in UCP. Active Sessions If a user is currently logged into UCP, you will see the IP address(es) of the active session(s). On the user's end, they will be required to login again to get back in. Select an extension by clicking in the field. A menu of available extensions will show up. Repeat the process if you would like to add multiple extensions. This is not the same as WebRTC. CDR Access Enter the extension(s) whose CDR data this user will be able to view. Allowed CEL Enter the extensions whose CEL this user will be able to view. Allowed Conference Bridges Select which conference bridge(s) the user will be able to access in UCP. Click in the field to display a drop-down list of available conferences. Click on a conference's name to select it. Repeat the process to select multiple conference bridges.

You can even create a “dead trunk” and route prohibited calls (such as international and 976 calls) to it. If you don’t add a pattern for international calls, then no one can call an international number. In this module, you’ll create an extension number and set a password for each extension and set-up voicemail (if desired). Note that this module only configures FreePBX and Asterisk. You must separately configure the phones themselves to connect to your PBX, either by configuring the phones manually, or by using the Endpoint Manager Module. You need the extension (user-id) and secret (password) for the phone to connect to the PBX. We recommend using the complex secret that’s automatically generated. Zulu UC is the application you need; it’s built to work with FreePBX and we have a comprehensive guide to setting it up. For example, you could make a Ring Group so that when any user dials extension 601, extensions 10, 11, 12, and 13 ring for 15 seconds, and then the call goes to the voicemail for extension 17. Something that has meaning to you). Most people send it to voicemail, but it’s up to you. This is the most complicated part. First you need to create recordings.Typically, you tell the PBX the phone number that outside callers have called (“DID Number” or “Direct Inward Dial Number”) and then indicate which extension, Ring Group, Voicemail, or other destination the call will go to. The Endpoint Manager module is provided for free with your FreePBXhosting.com hosting service. Customers are encouraged to use the backup and restore module which has a function to backup your FreePBX configuration files to an FTP site. Your system is now setup and you can begin to test. One of our staff members will update your ticket shortly with a detailed answer. Contact Sangoma Support for any questions regarding licensing or support terms. Free Month of FTP Hosting Use Promo Code: FreePBXFTP2015. Order an FreePBX Atom Dedicated Server and Receive 50% off for the first 5 months.

Use Promo Code: FreePBXATOM50off. Build a cloud phone system, a call center, a WebRTC application, or a custom audio service. This one-click build is ready to connect to your SIP phones and VoIP providers immediately. You can add extra functionality through free or commercial add-on modules. FreePBX and Asterisk are corporately sponsored by Sangoma Technologies. Simon Telephonics is providing this one-click install to the community and is not affiliated with Sangoma. Web Connect to the GUI at your droplet's IPv4 or IPv6 address. The initial form prompts you to set up your administrative login, so do this right away upon launch. After a series of setup prompts, including language selection and the enabling of the firewall (recommended), you will be presented with the FreePBX dashboard, a simple monitor and control panel for your PBX. Setup instructions including a getting-started guide and manuals for each module are available at the FreePBX Wiki. Documentation on the underlying Asterisk software can be found at the Asterisk Wiki. SSH SSH to your droplet's IPv4 or IPv6 address using the centos user (unprivileged, with sudo access) or root using your SSH key (preferred method). FreePBX command-line tasks can be accomplished using the fwconsole command; you can connect to the Asterisk CLI with asterisk -r once you have become root. You’ll need to either save your API access token to an environment variable or substitute it into the command below. All rights reserved. Proudly made in NY Solutions Agencies and Web Dev Shops Apps for Startups Apps for Working from Home Higher Education Kubernetes administration and operations Security and VPN Categories All Solutions Kubernetes Blogs and Forums Databases Developer Tools Frameworks Functions Monitoring For Vendors Getting Started Docs Support.

You can create multiple directories for your PBX, for example utilizing both Active Directory and local users that only exist on the PBX For more information on Directories see User Management Directories Settings The settings tab lets you define global User Manager settings Send Email on External New User Creation: Whether to send an email (using the 'Email Subject' and 'Email Body') to new users when they are created externally (not directly through User Manager) Send Email as HTML: Whether Email Body will send as HTML or plain text to the user Host Name: The hostname used for email. If left blank the default value of the address in your browser will be used Email Subject: Text to be used for the subject of the welcome email. Useable variables are: fname: First name lname: Last name brand: FreePBX title: title username: Username password: Password Email Body: Text to be used for the body of the welcome email Useable variables are: fname: First name lname: Last name brand: FreePBX title: title username: Username password: Password. For a better experience, please enable JavaScript in your browser before proceeding. It may not display this or other websites correctly. You should upgrade or use an alternative browser. I mean things like creating a new IVR, extensions, announcements, things like that. I'm willing to pay for a copy if some has a really good one they have put some time into. Has to be easy to understand for a novice administrator. Many Thanks Tim Am working on more too. Is there an updated link available? Sorry about that, many changes since then. Sorry about that, many changes since then.There is reference to a user guide below that, but I can't figure out how to download the user guide. Could it be your cart is misconfigured? By continuing to use this site, you are consenting to our use of cookies.

These range from the FreePBX appliance to the Below you can find some of the guides and manuals You will also find several Owning both End Point Manager Admin User Guide. Sangoma FreePBX FaxPro Brochure. Sangoma FeePBX Web Call Back Brochure. Sangoma EndPoint Manager Brochure Sangoma. PBXact PBXtended Features. Sangoma Expansion Module Datasheet Sangoma. PBXact Appliance with GXP21xx Phones User Manual Sangoma biult a solid reputation with. Below you will find links to tutorials, Getting Started Guides, Support Information and links to our partner sites for services that you might find useful. If you have additional questions please contact us. You will need to click APPLY CHANGES to actually apply the changes to the PBX. We suggest reading through this whole process before beginning. You need to click that to actually apply your changes to the PBX. FreePBX and Asterisk are powerful software packages but the setup can get rather complicated at times. If you find that configuring your PBX yourself is beyond your current ability you can always contact FreePBX.org for configuration support or consider upgrading to our fully managed PBX offering. Please read our guide on how to update your FreePBX server for details. This process will enable and configure the responsive firewall. All your phones go offline at once. Did you type an incorrect password a few too many times. Are you accessing FreePBX from a new location. If so, you might be blocked either passively by the firewall, or actively by Intrusion Detection. Don’t worry, it happens to all of us from time to time. Read our article on how to deal with this issue and prevent it going forward. A simple installation will tell the PBX to send all calls to a single trunk. However, a complex setup will have an outbound route for emergency calls, another outbound route for local calls, another for long distance calls, and perhaps even another for international calls.