PBX - Phone System User Guide - Wiki

Introduction

Why is your (PBX) phone system guide a wiki?

We’re international and encounter many different perspectives when it comes to phone systems. We’d like the guide to reflect that.

An American refers to it as a “call menu” while businesses in France may go with “IVR”. Somehow both should show up throughout the guide. Another example would be if someone has a better way of explaining a complicated forwarding scenario.

Any member with the proper “trust” level can edit. At the moment we’re not seeing any reasons for outright deletions, so if you’d like to remove a line or two, please replace it.

If you’re curious about signing up for our phone system, here’s an excerpt from one of our blog posts.

The grid itself invites creativity, you open up your panel and the first thing you see is a nice, clean, fresh grid ready for creating. (technically speaking it’s a “dot grid”) The changes are live, meaning if you attach your Brazil number [module] to the voicemail module- it will work right away.

Feature List

Below are available modules for the cloud PBX. They’re customizable and can be moved anywhere within the call flow (as long as the logic allows)

  • Internal numbers - aka extension numbers.

  • Ring groups - Send calls to multiple places, at once, or one after another among other options.

  • Voice menus - aka IVR or Call Menu. Route calls based on keypresses. (Press 1 for Sales, 2 for support, 3 for Vm etc. )

  • Audio playback - An audio file you can place within the call flow. Upload or record one.

  • Conference calling - Connect multiple people to have a conference call

  • Voicemail - Send calls to a voicemail box, send the VM’s themselves to an email, dropbox, or FTP.

  • Call recording​ - Record incoming and outgoing phone calls, send files to an email address or dropbox, Google drive, etc. Check your local laws in regards to recording phone calls.

  • Caller blocklist - aka spam filter. Block phone calls based on phone number, if it has a restricted caller ID, or even by area code/prefix.

  • Call queuing - aka “On-hold” - As callers come in, set them to ring “contacts” using round robin among other options.

  • Time routing​ - Route phone calls based on day and time. For example, set the calls to go to a certain voicemail box on weekends.

  • Caller routing​ - incoming calls are forwarded to different destinations, depending on the originating phone number.

  • Receive faxes - Convert incoming faxes to PDF, with multiple file delivery options available.

  • Event notifications - users receive alerts via email when specified events occur.

  • Configurable feature codes​​ - for accessing system functions such as call transfers and call pickup directly from the phone.

  • Call forwarding options​ - forward incoming calls to any phone number or VoIP destination.

The following are features catered to giving you, the user, a pleasant experience while setting up and managing the phone system.

  • Graphical user interface - call flows are configured via an easy-to-use, intuitive, drag-and-drop graphical web interface that is compatible with desktop computers, tablets and mobile devices.

  • Remote management - system management is achieved via a web-based interface, accessible from anywhere in the world.

  • Instant activation - voice configurations are instantaneously activated as they are graphically assembled, providing immediate access to the voice system.

  • Built-in contact center - for managing contacts and contact methods used by the phone system.

  • Built-in media center - for uploading audio files, recording messages and managing playlists that are used by the phone system

  • Receive calls on one or many phone numbers - add phone numbers used for inbound calling directly from the management interface.

  • SIP trunk configuration​ - Add and configure inbound and outbound SIP trunks directly from the management interface.

  • Call logs - access to detailed call history, with optional date filters.

  • Call statistics - access to detailed call statistics and charts.

  • Interconnect multiple instances of the phone system - unify remote offices into a single phone network with free internal calls using the secure Interlink​ feature.

  • Multi-language support - for global applications of the phone system.

  • Third-party compatibility - the phone system platform is specifically designed to seamlessly interconnect with any SIP-standard compatible VoIP service provider, hardware or software.

Overview

The phone system is quickly and easily configured via a graphical web interface, with drag-and-drop modules being connected together to define the call flows and the functionality of the PBX.

The components used in building the phone system applications are as follows:

  • Module - there are a number of different modules, with each module performing a specified PBX function such as voicemail, time routing or conferencing.

  • Module Menu - a menu listing the various modules that are used in building the call flow.

  • Settings Menu - a menu listing options for managing important supplementary phone system components such as phone numbers, internal numbers, contacts, media, inbound and outbound trunks, file delivery methods, feature codes, Interlinks and for accessing call logs.

  • Workspace - an area where the modules are placed and the call flows assembled.

  • Workspace Tab Menu - for displaying different pages of the PBX workspace.

  • Trash Bin - for removing modules from the workspace.

  • Cables - used to logically connect modules together to define call flows.

The phone system configuration screen consists of the following main components: the center Workspace where the PBX logic is assembled, the module Menu on the right-hand side of the screen, the Workspace Tab Menu along the top of the screen, and Trash Bin (when activated) in the lower right-hand corner.

Workspace

The workspace is used to assemble call flows. modules are dragged from the module Menu onto the workspace, configured, and then logically connected via “cables” to build the required voice system.

To select a module, position the mouse over the required module in the menu. Drag that module from the menu over the workspace, and release it where required.

Once modules have been positioned on the workspace, a configuration dialog box will be automatically opened on the right-hand side of the workspace. All required fields must be completed, and then the module is saved onto the workspace by pressing the save-button button.

modules that have been positioned on the workspace may be dragged and re-positioned as required.

Multiple Grids (Tabs)

The phone system allows the user to segment a voice system into logical groups and functions that may be arranged over multiple workspace pages. This feature is very useful when building complex voice systems, such as a PBX for a multi-branch business. The various workspace pages are accessed via the tabs on the Workspace Tab Menu , and tabs may be added, deleted or repositioned as required. In addition, the tabs may be labeled so as to define the functionality of each workspace page. Note that when the phone system is initially activated, there is a single tab denoted as Default . This tab may be renamed, but cannot be deleted unless at least one additional tab has been added to the workspace.

A new workspace page may be added by clicking on the add-tab-button on the top right of the grid.

A window is opened in which the name of the new tab must be entered, and the Tab Menu is updated by clicking on the save button.

A new, blank workspace is created, and by using this example, is accessed by selecting the “Sales” tab.

Sales tab

Placing the mouse over a tab name displays image icon. The name of the tab may be changed by clicking on the icon, which opens a tab configuration window. Note that a tab and its associated workspace page may deleted by clicking on the image , even if there are PBX modules on that page. With that being said, please be careful when deleting tabs.

drag tabs

Tab positions may be changed within the Workspace Tab Menu by “dragging” tags horizontally along the tag listing bar.

The phone system allows users to segment voice systems into logical groups and functions that may be arranged over multiple workspace pages. This feature is very useful when building complex voice systems, such as a virtual PBX for a multi-branch business.

Moving Modules between Workspace Pages

Modules may be moved between workspace pages. In the illustration below, two workspace pages are available: Default and Sales , with the Default workspace comprising a number of modules, including an module “Sales Voice Menu”. Note that the name of the active workspace page (i.e., the page that is currently being displayed) is shown in blue.

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If it is required that the voice system logic for the sales division should be assembled on the Sales workspace page, then the “Sales Voice Menu” module should be moved to that page. To achieve this, “drag” the “Sales Voice Menu” module over the Sales workspace tab, until that tab is shown in blue to indicate that the Sales workspace is active. Do not release the module.

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Continue “dragging” the “Sales Voice Menu” module downwards into the active Sales workspace, and release that module in the desired position on that workspace. Further assembly of the voice system logic for the sales division may now continue on this workspace.

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Note that re-selecting the Default workspace tab shows the modules remaining on this page. The “Sales Voice Menu” module is no longer present, as it has been moved to the Sales workspace page.

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Connecting cables between modules on different workspace pages

Partial voice scenarios assembled on different workspace pages must be interconnected in order to build complete voice systems. In the example above, calls must be passed from the “Main Voice Menu” on the Default workspace page to the “Sales Voice Menu” on the Sales workspace page.

To generate a cable between these two modules, “drag” a cable from the right-hand socket of the “Main Voice Menu” module, placing the mouse pointer over the Sales workspace tab, until that tab is shown in blue to indicate that the Sales workspace is active. Do not release the cable.

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Continue “dragging” the cable downwards into the active Sales workspace, and position that cable over the left-hand socket of the “Sales Voice Menu” module. Release the cable.

The cable connection between the “Main Voice Menu” on the Default workspace page and the “Sales Voice Menu” on the Sales workspace page is now complete.

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The logical connectivity between modules on different workspace pages is represented by a cable terminating in a bubble icon. By clicking on this bubble icon, the alternate workspace page associated with this connection will be displayed.

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Note that the cable termination icon bubble may be dragged and repositioned independently on each individual workspace page, without affecting the position of the logically connected termination icon on the other workspace page.

Important - if two modules on the same workspace page are connected, then moving one of the modules to a different workspace page will not disconnect the modules. Instead, these modules will remain connected, with the bubble icon indicating the cable continuity over the different workspace pages.

Example scenarios using multiple pages

In the PBX configuration below, the main voice menu directs incoming calls to either a sales or support departmental voice menu, and then these calls are further distributed to members within each department.

This PBX configuration may be logically divided into three separate functional groups:

  • The main PBX logic consisting of the incoming phone numbers and the main voice menu.
  • The PBX logic for the sales division (starting with the sales Voice Menu module).
  • The PBX logic for the support division (starting with the support Voice Menu module).

In order to segment this voice scenario, additional workspace pages must be added and workspace tabs are labeled Main, Sales and Support.

In order to logically segment this voice system, modules on the Main workspace page may now be moved to other workspace pages as detailed in the section Moving modules between Workspace Pages above.

IMPORTANT: Instead of assembling the complete voice system on a single workspace page and then moving modules to other pages, it is easier and more practical to first create the required workspace pages and assemble the voice “sub-systems” on each of those pages.

Once this has been done, then the partial voice scenarios on different workspace pages may be interconnected as described in the section Connecting Cables Between Modules on Different Workspaces Pages.

Referring to the voice scenario described above, the logical segmentation results in three separate workspaces pages as follows:

The Main workspace

The Sales workspace

The Support workspace

Settings Menu

The Settings menu is accessed by clicking on the hamburger icon image on the top-right-hand corner of the screen.

The available settings are as follows:

phone icon Phone Numbers - Add and manage phone numbers that are used for inbound calling and for displaying as caller IDs when making outbound calls on SIP devices. In addition, a list of Phone Numbers together with their associated modules may be exported. See the section Phone Numbers for further details.

internal-ext Internal numbers (i.e. Extensions) Add new Internal Numbers (ext.) and display a list of all the currently configured Internal Numbers, together with the names of the modules that these numbers are connected to. In addition, a list of Internal Numbers together with their associated modules may be exported. See the section Internal Numbers for more info.

contact icon Contact Center - Add and edit contacts and contact methods that will be used by the various phone system modules. See the section Contact Center for further details.

media icon Media Center - Upload files, record messages and create, modify and delete playlists that are used by the phone system . Playlists consist of pre-recorded music, commercials or any other messages that may be played to the caller while they wait for their call to be answered.

delivery icon Delivery Methods - list and configure methods that may be used for the delivery of audio and text files generated by the phone system. These files include voicemail and notification messages, as well as the contents of recorded calls. Delivery method options are Email, Dropbox, FTP, SFTP, Google Drive and OneDrive.

call log icon Call Logs - Display CDRs (Call Detail Records) and service usage charts such as call distribution and lost call rates. CDRs may be selected according to the call direction (inbound or outbound), and filters such as the source and destination phone numbers, gateway, call contact or date range may be applied. See the section [Call Logs, Charts and Statistics]

feature code icon Feature Codes - Configure phone system feature codes such as required for DTMF transfers, call pickup and call recording on demand.

general icon General Settings - Select the language for the phone system menus, with the current options being English, Russian, Lithuanian and Latvian. Also, set the system time zone, choose the decline code that is sent to the caller when maximum channel capacity is reached, and enable or disable the dark mode for the user interface.

To close the Settings Menu, click on the image icon in the top right-hand corner of this menu.

Module Menu

You’ll likely want to start here, here’s where you’ll be able to drag a module right onto the canvas grid.

This menu contains the various modules that may be used in setting up the PBX. This menu can be minimized to create a larger workspace by clicking on the image icon located at the bottom right-hand corner of the workspace.

Modules

There are a number of different modules (also called modules), each performing a specific function or set of functions. These modules can be arranged and inter-connected in a wide variety of combinations, with calls being passed from one module to another as required.

Modules are dragged from the module Menu onto the workspace, and once an module has been added to the workspace, it must be configured. Configuration options for all modules are specified in the section module Configuration.

Each module has either one or two sockets which are shown as small protrusions on the left and/or right-hand sides of the module, and these sockets are used for connecting the modules together via cables. The left-hand socket acts as the input to an module, while the right-hand socket acts as the output from that module

modules or modules

Duplicating modules

The phone system includes the ability for users to duplicate modules on the workspace. All configuration parameters in the original module are copied to the new module, however the name of the new module is modified to include a copy number. For example, the first duplication of a Voice Menu module named “Voicemail” will be named “Voicemail Copy 1”.

To use this copy feature:

  • For MacOS, hold down the “option” key when dragging an module.

  • For Windows, hold down the “Ctrl” key when dragging an module.

It is important to note that not all modules may be duplicated. For example, Phone Number and Internal Number modules are assigned unique phone/internal numbers and may therefore not be copied.

Moving multiple modules

The phone system allows users to select multiple modules and reposition them simultaneously on the workspace, or move all of the selected modules to another tab. Once an module has been selected, it will be marked with a blue border.

To use the module selection feature:

  • MacOS - hold the ‘command’ key to select individual modules

  • MacOS - hold the ‘shift’ to select a complete module tree (modules connected by cables)

  • Windows - hold the ‘ctrl’ key to select individual modules

  • Windows - hold the ‘shift’ to select a complete module tree (modules connected by the cables)

Use the same keys as listed above to deselect modules, or to exclude particular modules from the module tree selection. Clicking on an empty area on the workspace will deselect all previously selected modules or module trees. Additionally, selected module trees may be moved to the Trash Bin.

Cables

Cables are used to logically interconnect the modules that have been placed in the workspace area, thereby defining the call flows and the functionality of the PBX.

To create a cable, place the mouse over the right-hand socket of a module, and use the mouse to “drag” a cable from that socket towards the left-hand socket of the destination module.

connect cables

Once the end of the cable is over the left-hand socket of the destination module, release the cable and the two modules will be logically connected as required.

connected cables

The system includes an intelligent cabling configuration assistant that serves to simplify the connection logic between modules by indicating possible cable attachment points. In the figure below, once a cable has been generated from the Phone Number module/modules, all valid input connection options on the workspace are shown by the available modules having their left-hand sockets colored blue. The cable may be connected to any one of these sockets.

available connections

To remove a cable and logically disconnect two modules, place the mouse pointer over that cable until the “delete cable” icon image appears. Click on that icon to complete the removal of the cable.

remove connection

A “Delete Connection” window will appear, displaying the details of the module connection to be terminated. To complete the action, click the “Delete Anyway” button, or select the image icon to leave the connection unchanged.

confirm delete

In some cases, multiple cables, each having a distinct logical function, may be generated from the exit (right-hand) socket of an module, and these cables are connected to various modules as required by the call flow. For example, when configuring a voice menu, there are three different logical options for the cables that are generated from the right-hand side of this module:

  • Calls are forwarded to modules according to the extension number entered by the caller.

  • Calls are forwarded to a specific module if the caller enters an invalid extension number.

  • Calls are forwarded to a specific module if the caller does not enter an extension number within a defined timeout period.

When a variety of logical functional options may be assigned to a single cable, then once the cable has been connected between two modules, a configuration window is automatically displayed that allows the user to select the required function for that cable.

The screenshot below illustrates the functionality of multiple cables exiting a Voice Menu module. After the voice message has been played to the caller, if the caller presses “100” then the call will be forwarded to the sales ring group, and if “200” is pressed, then the call will be forwarded to the support ring group.

Invalid extension and timeout conditions (denoted as i and t respectively on the cables) are forwarded to specified Audio Playback modules where appropriate messages are played to the caller.

Note that the allocated extension numbers may be changed by clicking on the configured number displayed on the cable. A configuration dialog window is opened, and a new extension number may be entered.

Similarly, some modules, such as Time Router or Caller Router, have two right-hand (exit) sockets, which are used to implement a “Yes/No” call flow logic. For example, for a Time Routing module, a cable generated from the “Yes” (green) socket defines the routing of calls if those calls are received within the configured day/time interval, and the “No” (red socket) option defines the routing in the case of a time period match failure.

In the illustration below using a Time Router module, if the incoming call is received within the configured day/time parameters, then the call will be forwarded to Sales. Otherwise, the call will be forwarded to an Audio Playback module where a pre-recorded “after-hours” message is played to the caller.

time router

It is important to note that if a cable is not connected from the left-hand (output) socket of a module to the right-hand (input) socket of another module, then a call will be terminated if the PBX logic attempts to pass that call to adjacent modules.

For example, in the illustration below, an incoming call is forwarded to an Audio Playback module where a message is played to the caller. Because the right-hand side of the Audio Playback module does not have a cable connected to another module, the call will be terminated as soon as that message has been played.

phone to VM modules

Trash Bin

The Trash bin allows the user to delete modules that have been previously placed on the workspace. To delete an module, drag that module towards the plus or the x icon at the bottom right-hand corner of the workspace. This icon will be replaced by the trash bin icon trash icon, and the module to be deleted should be dragged and dropped over the Trash Bin.

Note that the trash bin does not have a recovery feature, anything that goes in it is gone forever.

Search

The search tool is used to find related phone system resources with a single search request. It is effectively a combination of the existing search capabilities, with the addition of the ability to search for associated modules, contacts, delivery methods and phone numbers.

To begin the search, click on the search icon in the Workspace tab menu to activate the search window.

In the search field, type in the name of the resource to be located and click on the Search button. Once the search has been completed, all results will be displayed and grouped by resource type.

The button image may be used to expand and view the grouped search results. Additional buttons image and image provides the option to locate the module on the workspace or to access the resource’s settings window.

Using Touchscreens

The phone system user interface is compatible with touchscreen-enabled devices. The major differences between regular and touchscreen usage are as follows:

  • For a touchscreen, to select an module that should be moved on the workspace or that requires the connection/deletion of an output cable, apply a long press (long tap) to that module.

This action results in the touch icon being displayed above the selected module, and also causes the output socket of the module to be highlighted. The module may now be moved, or a cable may be connected between that module and another module on the workspace as detailed below.

Moving an module - Once the touch icon is displayed, the module may be “dragged” and repositioned on the workspace. Tapping on a blank area in the workspace removes the touch icon.

Connecting modules with cables - Once the output socket of an module is highlighted, a second short tap on that socket causes the output socket of the selected module and the valid input sockets of all other modules on the workspace to be highlighted with pulsating blue circles.

A cable may now be dragged from the output socket of the selected module, and connected to the input socket of a destination module.

Again, tapping on a blank area in the workspace removes the touch icon.

Deleting a cable - Once the touch icon is displayed, the cable connecting the selected module to another module on the workspace is displayed, with the “delete cable” icon appearing in the center of the cable. Clicking on the red x icon icon removes the cable.

Phone Numbers

Where the phone call starts.

In the hamburger settings menu image, you’ll find the “Phone Numbers” image setting.

Above is where you can manage the phone numbers as well as click the “map” icon to find your numbers on the grid. ( Especially helpful if you have multiple FlyNumbers)

Phone numbers are used for:

  • Configuring the phone number module and being the phone number to be dialed for inbound calls.

  • Select the caller ID to be displayed when making outbound calls from VoIP/SIP devices.

The Phone Numbers window includes two tabs; a “List” tab for managing the Phone Numbers

There’s also a “Directory” tab so you can see what all your numbers are “doing”.

Selecting the “Directory” tab displays a list of numbers allocated to configured Phone Number modules, together with the name of the modules on the workspace to which those Phone Number modules have been logically connected. This directory may be optionally exported in .csv format by clicking on the exportbutton.

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Internal Numbers (Extensions)

From the hamburger menu (Top-right) - settings , you can open the image configuration screen.

This is to manage your extensions or as labeled on our PBX - “Internal numbers”.

The Internal Numbers window includes two tabs; a “List” tab for managing the Internal Numbers, and a “Directory” tab for displaying the mapping between the configured numbers and modules on the workspace, with an option to export that directory.

When the “List” tab is selected, new Internal Numbers are added by clicking on the add new number option, with numbers being from 1 to 4 digits in length.

These numbers may be allocated to Internal Number modules as and when those modules are added to the phone system workspace.

To change an Internal Number, the gear icon icon should be selected.

Clicking on the map icon icon closes the Internal Numbers window, and the module associated with that number oscillates for a short duration on the workspace, making it easily identifiable.

Internal numbers that are no longer required may be deleted by clicking on the red x icon to the right of the number. Note that numbers currently in use by the PBX logic and allocated to Internal Number modules may not be deleted.

Selecting the “Directory” tab displays a list of numbers allocated to Internal Number modules, together with the name of the modules on the workspace to which the Internal Numbers modules have been logically connected.

This directory may be exported in .csv format by clicking on the export button.

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Contact Center

The Contact Center manages the contacts and contact methods used by the phone system. Specifically, the entries in the Contact Center serve to list the destinations (such as people and departments) to which incoming calls are redirected, and also the methods (such as landline, mobile phones or SIP Accounts) by which these destinations are contacted.

The Contact Center is accessed by clicking on the image hamburger menu in the upper right-hand corner.

This opens the Settings menu, which includes the “Contact Center” option.

It should be noted that contacts and contact methods may be pre-configured in the Contact Center before adding modules such as Ring Groups (which require these contacts for call forwarding) to the workspace.
However, the Contact Center is made available to users during module configuration, and contacts/contact methods may be added and edited at that time.

There are two components for each contact:

  • The actual contact, such as a person or a department of an enterprise. The contact is identified by a first and last name.

  • The contact method/s associated with the contact. Multiple contact methods may be allocated to each contact, and the options are:

    • Phone Number - Lists the PSTN phone number (landline or mobile) to which inbound calls may be forwarded.

    • SIP Forwarding - Allows the forwarding of calls to a SIP network, such as that of an ITSP.

    • SIP Account - Enables the use of a VoIP compatible end-device, such as a softphone. SIP credentials such as username, password and domain are provided by the phone system for configuration of the SIP device. The SIP Account contact method supports internal dialing and outbound calling, as well as optional call recording.

    • Email - Used by modules such as Voicemail to specify the email address to which files containing voice messages must be sent.

  • Each contact method includes a Label (for example, Work, Home, Office, Mobile) that serves to identify the contact method, as well as the required configuration details for that particular contact method.

A typical list of contacts is shown below, together with the number of contact methods configured for each contact. Each configured contact and their contact methods may be edited by clicking on the gear icon icon next to that contact , and new contacts may be added by selecting the “Add new contact” option.

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Create Contact

Selecting the option “Add new contact” within the Contact Center window opens a form for creating a new contact. There is an option to include this Contact in the company directory (Show in Company Directory) which is enabled by default. The first and last names for the contact must be entered, and optional details for the fields Company, Job Title and Contact Numbers may be entered. It is not required to create a dedicated Sip-Account for each call.center device, as this can be replaced by the automatically created call.center contact method.

However, if needed, additional SIP-Accounts may be added to each call.center device. Contacts that have the company directory feature enabled will be provisioned to the call.center application and are updated/synchronised automatically. The call.center contact method may be used with Ring Groups and Queues directly.

Note: Contact numbers will be used when calling the contact from call.center’s company directory.

By selecting the “Add new number” option, you can enter an internal number or phone number (landline or mobile) for the contact. Phone numbers are in E.164 format:

<CountryCode> <City/AreaCode> <LocalNumber>

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After a new contact has been added, an avatar may be selected for that contact.

Note: A default avatar based on the contact’s name and surname will be automatically allocated to the contact.

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To change the avatar, click on the gear icon icon to open up the settings window for the selected contact.

Click on the cam icon icon at the top of the window to open up the native file explorer, and select the image you would like to use as the avatar for the contact.

image

After the picture has been selected, an avatar editor window will open for the optional resizing and cropping of the image. Click on the Save button once the editing has been completed.

Configure Contact Method

Or in other words, where to send the calls for this “Contact”

In the Contact Center, select the contact from the list of configured contacts. Click on the gear icon icon next to the contact to open the contact details.

Select the “Add new contact method” option within the contact window to open a menu for adding a new contact method.

Once the required contact method has been selected from the drop-down list, configuration details for that particular contact method must be entered.

The configuration options for each “contact method” are as follows:

Regular Phone Number

Using this method has a low per min rate depending on the forwarding number. For example, sending the calls to a US or Canada phone number would be an additional .01. Unlimited plans coming soon.

The available settings are as follows:

  • A label (or identifier) for this Phone Number contact method , for example, “Work”. An existing label may be chosen, or a new label may be defined by selecting “Add a new label”

  • The actual phone number (landline or mobile) to which incoming calls should be forwarded, in E.164 format:

<CountryCode> <City/AreaCode> <LocalNumber>

The country code is 1-3 digits long, while the length of the city/area code and local number may vary. An example of a phone number in E.164 is 17185551212for NY, US.

SIP Forwarding

Send the calls to a SIP URI

The available settings are as follows:

  • A label (or identifier) for this SIP Forwarding contact method, for example, “John SIP”. An existing label may be chosen, or a new label may be defined by selecting “Add new label”

  • The username, domain and/or port number for the SIP service, obtained from the service provider or system administrator.

An example would be if you want to send the calls to an Asterisk with EXT 200

“Username” would be 200 and “Domain” would be the Asterisk itself, 1.1.1.1 or john.asteriskserver.com for instance.

Create SIP Account

Here the phone system will provide you SIP credentials which you can enter into any SIP app or device.

There are three tabs in the SIP Account configuration window, one for configuring the actual SIP Account parameters, another to optionally enable call recording, and the third to select outbound call announcements and advanced settings for this SIP Account .

SIP Account Main Tab

  • A label (or identifier) for this SIP Account contact method, for example, “Mike’s-Softphone”. An existing label may be chosen, or a new label may be defined by selecting " Add new label".

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  • The caller ID name, which will be displayed on end devices that support this feature. By default, this name is the same as the actual contact name to which this contact method is attributed.

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  • An optional Internal caller ID number may be selected from the dropdown menu of the extensions configured for Internal Number modules that are currently on the workspace. This number will be displayed as the Caller ID when making outbound calls to other extensions within the phone system network.

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  • An option to enable outbound calling from this SIP device. By default, outbound calling is disabled.

  • If outbound calling is enabled, an external caller ID must be selected that defines the phone number to be displayed as the Caller ID when making outbound calls. This Caller ID may be chosen from the dropdown menu that lists numbers previously added to the phone system by using the Phone Number option under the “Settings” menu. Therefore, it is recommended that phone numbers should be added prior to configuring SIP Account contact methods.

Note: The dropdown list of internal numbers that is displayed consists of numbers (or extensions) allocated to configured Internal Number modules. Therefore it is recommended that Internal Number modules be configured and placed on the workspace prior to adding SIP Account contact methods.

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SIP Account Call Recording

The SIP Account contact method (and also the External Line contact method if applicable) provides users with the option of enabling call recording, including the ability to define the recording direction (inbound and/or outbound) and to record internal and/or external calls. In addition, a “record on demand” feature is available, where the user may dial a predefined feature code to activate call recording.

Note that by default, call recording is disabled.

If call recording is enabled, then the delivery method for the file containing the contents of the recorded call must be selected, with the options being Email, Dropbox, FTP, SFTP, Google Drive, OneDrive. If the required delivery method has not previously been configured by using the Delivery Methods option listed under the “Settings” menu, a delivery method may be added by selecting the “Add deleviery method” option in the dropdown menu.

If the “record on demand” feature is enabled, then a dialing feature code must be defined in order to activate call recording.

Feature codes are configured by selecting the image option under the Settings hamburger menu .

SIP Account Advanced Tab

Whitelist IP’s for your SIP credentials.

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  • The “Enable Allowed IPs” option, which if activated, is used to limit device registration to specific, listed IP addresses.

This option prevents possible security violations or fraud by disallowing device registration from unlisted IP addresses. If the “Enable Allowed IPs” option is not enabled, then registration from all IP addresses is permitted.

  • If the “Enable Allowed IPs” option is enabled, then the “Allowed IPs” list must include the IP addresses to which device registration is limited. Both IPv4 and IPv6 addresses may be listed, with the use of the “SPACE” character indicating the completion of an address. In addition, an IP address range may be defined using the standard CIDR (Classless Inter-Domain Routing) “/” subnet mask notation.

  • The codecs to be supported by this SIP Account. Multiple codecs may be added from the dropdown list, and the options are:

    • OPUS
    • G722
    • PCMU
    • PCMA
    • G729
    • GSM
    • telephone-event
  • If multiple codecs are listed, then codec priorities may be set by dragging the included codecs into their desired positions of preference.Important - the telephone-event codec must be included in the list of “Allowed Codecs” if interactive menus or feature codes requiring the input of digits are to be used.

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The “Media types” field allows the selection of the network protocol used for delivering audio and video over IP networks.

The following options are available:

  • RTP (Unsecured Real-time Transport Protocol)

  • SRTP-SDES (Secure Real-time Transport Protocol - Session Description Protocol Security Descriptions)

  • SRTP-DTLS (Secure Real-time Transport Protocol - Datagram Transport Layer Security)

The “Transport protocol” field allows for the selection of protocols used for communication between the phone system and end-user’s IP Phone or softphone application.

Multiple transport protocols may be selected, with the following options being available:

  • TCP (Transmission Control Protocol)
  • UDP (User Datagram Protocol)
  • TLS (Transport Layer Security)
  • WSS (WebSocket Secure)

Note that encrypted media types and transport protocols are disabled by default. Contact us for more details on enabling on your account.

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The SIP Account contact method provides users with the option of enabling outbound call announcements, with the announcement being made to the operator once the call has been connected with the end point. Users may set an outbound call announcement for both internal and external calls.

When using this feature, the caller will hear on hold music and the end point will hear the selected announcement. To enable this feature, click on the “Internal/External outbound call announcement” section in the dropdown menu and select the relevant audio file for the announcement.

On completion of the configuration of a SIP Account contact method , the SIP credentials may be viewed by expanding the SIP details window.

SIP Account Credentials

There are two ways to find a contact in the Contact Center. You may select the contact directly from the displayed list of configured contacts, or by using the search option. Please refer to the user guide section Search for further details on how to use the “Search” option.

Click on the gear icon icon next to the contact to open the Contact Details.

Click the gear gear icon icon next to the selected contact to open a SIP Account configuration window.


Clicking on the down icon icon will expand the SIP details window.

This window includes the SIP credentials (such as username, password and domain) automatically generated by the phone system and are required for configuring a SIP device. In addition, the current status of the SIP device (Offline or Online ) is shown.

Clicking on the copy icon icon next to the Username, Password and Domain fields copies the configuration data to a clipboard for convenient use in configuring the SIP device.

Module Configuration

Once a module has been dragged from the menu and released onto the workspace, a configuration dialog box is automatically opened.

(Below is what happens when you drag the conference module onto the grid)

Each module has a specific set of configuration requirements, and the module will only be usable once the configuration has been successfully completed and the save-icon option selected.

If you choose not to configure the module but instead select the x icon icon at the top right-hand corner of the configuration window, then that module will be removed from the workspace.

Once a module has been configured, then clicking on that module will display selected configuration details. For example, clicking on the Conference module expands that module, showing the required participation PIN code.

click module for info

module details

You can always update the configuration settings for any module/module by clicking the image gear icon on the top right of the module.

Below is a breakdown for each module.

Phone Number (Module)

The “Phone Number” module. Usually the first part of your call flow.

Phone number module

Configuration options for the phone number module.

Your FlyNumbers will automatically show up as phone numbers and you can select them when prompted (or clicking by the gear icon in the module) image

  • The name of the Phone Number module.

This helps in identifying the number on the workspace, for example, “Sales team NY”. Note that newly created Phone Number modules are provided with default names in the format {Tab name} {module type} {module type count in current tab}, for example “Main Phone Number 3”.

  • The phone number, which may be selected from a list of unused phone numbers that FlyNumber automatically propagates for you.

A phone number may only be allocated to a single Phone Number module, and may not be reused by other such modules.

CLI Rules are used to override the Source Caller Name. This functionality allows flexible Caller Name configurations and may be used to differentiate SIP calls you receive from the phone system. CLI Rules will help you to identify from which Phone Number module the call is coming from.

Note: By default, the CLI Rules feature is disabled.

Extensions (Internal numbers)

The Internal Number module is used for internal dialing as an extension number. This facility allows users to call each other directly and to reach selected modules (such as Conference and Voice Menu modules) via internally-assigned extension numbers.

Please note that one Internal Number module must be created for each internal extension required, and extension numbers cannot be duplicated on multiple Internal Number modules.

ext module

Configuration options are as follows:

  • The name of the Internal Number module, for example, “Sales Conf Extension”. Note that newly created Internal Number modules are provided with default names in the format {Tab name} {module type} {module type count in current tab}, for example “Main Internal Number 7”.

  • An internal number (extension number) that will be used for internal dialing (1 to 4 digits). This number may be selected from a list of unused internal numbers that were previously added via the option image in the “Settings” menu.

A simple usage example of the Internal Number module is shown below, where an internal extension number is dialed by a phone system user to access the company’s conferencing facility.

ext example

Ring Group

The ring group module redirects incoming external or internal calls to different destinations included in the ring (or hunt) group. These destinations consist of contact methods, such as phone numbers (landline or mobile) and VoIP connections that are configured for the various contacts in the system.

Multiple contacts and contact methods may be included as call destinations within a single Ring Group module, and the ring times and ring sequences for the selected contacts/contact methods are fully configurable.

If the first contact/contact method is busy or remains unanswered for the set time period, the call is passed to the next device in the configured ring sequence, and so on through the list of contact methods. Alternatively, all destinations in the ring group may be configured to ring simultaneously.

The contacts and their various contact methods to be used by Ring Group modules may be either configured when assembling the ring group, or may be pre-configured using the Contact Center found under the Settings menu (see the section Contact Center for further details).

The Ring Group module includes the ability to configure playlists as both “music-on-hold” and “ringback-tone”, so that pre-recorded music, messages, commercials or any other audio clips may be played to callers.

ring group

The configuration settings for ring group module are as follows:

  • The name of the Ring Group module, for example “Sales Ring Group”.

  • Contacts, defining a ring destination or multiple ring destinations to which incoming calls will be forwarded. Each ring destination consists of a contact and an associated contact method, and may be directly configured from the Ring Group module or may be pre-configured by using the Contact Center. The section Contact Center provides full details on how to configure contacts and contact methods.

Selecting image allows the user to add a ring destination from a drop-down menu of pre-configured contacts and their associated contact methods. Alternatively, a new contact may be added by selecting image.

Once the ring destinations have been added to the Ring Group module, the ring times and, if applicable, the ring sequences may be configured. The ring times for each contact method are shown on a 60-second timeline bar, and each contact method may be placed as required along the timeline bar.

This is achieved by positioning the mouse over the selected contact method, and then “dragging” the contact method along the timeline bar to the required position.

Note: At least one of the contact methods must have a ring time starting at the zero-second mark.

The ring start and end times for each contact method may be changed by “stretching” or “shrinking” that contact method. To do this, place the mouse over the left or right-hand edge of the contact method, and re-size that contact method by “dragging” the mouse in a left or right-hand direction as required.

Note: The position of the Contact Methods on the timeline bar are automatically adjusted to ensure that there are no time gaps between the end ring time of one contact method, and the start ring time of the next sequential contact method.

Media Tab - Ring Group

The Media tab is used to include media as both “music-on-hold” and “ringback tone” playlists. This allows pre-recorded music, messages, commercials or any other audio clips to be played to callers before their call is answered (ringback tone), or while an active call is put on hold (music-on-hold). The section Media Center provides full details on how to configure and manage media files.

It should be noted that playlists must be pre-configured in the Media Center before adding modules to the workspace such as Ring Groups, which may include the use of these audio files.

CLI Rules - Ring Group

The CLI Rules are used to override the Source Caller Name. This functionality allows flexible Caller Name configurations and can be used to differentiate SIP calls from the phone system. CLI Rules will help you to identify from which Ring Group module the call is coming from.

Note: By default, the CLI Rules feature is disabled.

The only cable exiting the Ring Group module is for a “timeout” condition, defining the path if an incoming call is not answered within the maximum configured ring time.

A simple usage example of the Ring Group module is shown below, where an incoming call is directed to the Sales Ring Group. If the call is unanswered within the maximum configured ring time, then that call is forwarded to voicemail.

Multiple Destinations - Ring Group

Selecting image allows the user to add a multiple ring destination from a drop-down menu of pre-configured contacts and their associated contact methods.

For example, you. can send the calls to a regular phone number, SIP device, and 2nd phone number using 1 ring group module.

Voice Menu / IVR

Also known as a “Call Menu” ( i.e. press 1 for sales, 2 for support, 3 for VM, etc. )

The Voice Menu module is used for implementing an IVR (Interactive Voice Response) or automated attendant system, allowing callers to listen to a recording and navigate to different destinations using their dial pad. This module acts as a virtual receptionist and includes the ability to play key messages and pass information to callers.

On receiving an incoming call, the Voice Menu plays an audio message, prompting the caller to enter an extension number. The call is then passed to the connecting module with the matching extension number. Logic is included so that erroneously entered extensions or caller-entry timeouts may be properly handled.

menu module

Configure Menu/IVR

The configuration options for the “voice menu” module are as follows.

  • The name of the Voice Menu module, for example, “Main Office Menu”.

  • An audio file that may be uploaded from a local drive (in .mp3/.wav/.flac/.ogg format), may be recorded directly, or may be selected from files/playlists previously uploaded into the phone system Media Center. Typically this audio message would include information pertaining to extension numbers that are to be entered by the caller on their dial pad in order to connect with people or departments.

A timeout (from 1 second to 2 minutes), defining the maximum time allowed for the caller to enter an extension number using their dial pad. This timer starts immediately after the audio message has been played. The timeout value is changed by positioning the mouse over the right-hand edge of the timeout bar, and then “dragging” the edge in a left or right direction.

  • A playback counter (from 1 to 11) allows the sequence of playing the selected audio file and completing the caller-input timeout to be repeated before that call is forwarded in accordance with the “Reached timeout limit” logic as described below. The playback count is changed by positioning the mouse over the right-hand edge of the “Playbacks” bar, and then “dragging” the edge in a left or right direction.

It is important to note that cables exiting from the right-hand socket of the Voice Menu module serve three possible functions:

  • Extension - The cable forwards incoming calls to the appropriate module in response to a valid extension number entered by the caller.

  • IVR invalid selection - The cable forwards incoming calls to a specified module (such as an Audio Playback module) if the caller enters an invalid extension number.

  • Reached the timeout limit - The cable forwards incoming calls to a specified module (such as an Audio Playback module) if the caller does not enter an extension number within the defined timeout period.

When a cable is generated from a Voice Menu module and is connected to another module, a configuration menu is automatically displayed, prompting the user to select the Connection type (“Extension”, “IVR invalid selection” or “Reached the timeout limit”) for that cable.

cable menu

If a Connection type “Extension” is selected, an extension number must be entered to match the instructions in the voice message that is played to the caller. This extension number will be displayed on the cable that connects the Voice Menu module to the adjacent module.

ext menu

In the usage example of the Voice Menu module shown below, a voice message is played to the caller, with the instructions “Press 100 for sales, and 200 for support". If the caller presses “100”, then the call will be forwarded to the sales ring group, and if “200” is pressed, then the call will be forwarded to the support ring group. Invalid extension and timeout conditions are forwarded to specified Audio Playback modules where appropriate messages are played to the caller.

Extension numbers may be changed by clicking on the configured number displayed on the cable. A configuration dialog window will be opened, and a new extension number may be entered.

It should be noted that in the case where the caller enters an invalid extension number or a timeout occurs, the call may be “looped” back to the Voice Menu module. This will cause the instructions regarding valid extension numbers to be replayed to the caller, and the caller will have an additional opportunity to contact the desired party.

In the illustration below, calls generating error conditions are forwarded to Audio Playback modules where specified messages are played to the caller (for example, a message “You have entered an invalid extension number”). On completion of this audio playback, the call is returned to the Voice Menu module.

Audio Playback

Audio Playback allows an audio message such as a voice recording or music-on-hold to be played to the caller. After the audio file has been played, the call is passed to the module connected to the right-hand socket of the Audio Playback module.

playback module

The settings for the playback module are as follows

  • The name of the Audio Playback module.

  • An audio message that should be played to the caller. In general, this audio file will have previously been uploaded from a local drive or recorded, and stored in the phone system Media Center. New files may be added to the Media Center by selecting the image option in the “File” dropdown menu.

An example scenario involving the Audio Playback module is shown below, where a voice message is played to the caller, after which the call is passed to a Ring Group module (“Mike Brown”) for further processing.

Voicemail

The Voicemail modules serve as a mailbox in which callers may leave voice messages, which are then immediately sent to a specified destination on the termination of each call. We don’t store the files.

The configuration settings for the voicemail module are as follows

  • The name of the Voicemail module.

  • The delivery method for the voicemail audio message, with the options being Email, Dropbox, FTP, SFTP, Google Drive or OneDrive.

If the required delivery method has not previously been configured by using the Delivery Methods option listed under the “Settings” menu, a delivery method may be added by selecting the option image in the dropdown menu.

An audio message that should be played to the caller. Generally, this audio file will have previously been uploaded from a local drive or recorded, and stored in the phone system Media Center. However, new files may be added to the Media Center by selecting the image option in the “File” dropdown menu.

The maximum length of the voicemail, with valid values being from 1 to 60 minutes. The call will automatically be terminated after this configured time period, unless the caller hangs up before the expiration of this timer.

A simple usage example of the Voicemail module is shown below, where an incoming call is forwarded to the ring group Mike Brown. If Mike does not answer the call within the defined ring timeout period then that call is sent to voicemail, where the caller may leave a message.

Fax

The Fax module allows for incoming faxes to be stored in PDF format, with multiple file delivery options being available. G.711 pass-through and T.38 fax protocols are supported.

fax module

The settings for the fax module are as follows

  • The name of the Fax module.

  • The delivery method for the fax PDF file, with the options being Email, Dropbox, FTP, SFTP, Google Drive or OneDrive. If the required delivery method has not previously been configured by using the Delivery Methods option listed under the “Settings” menu, a delivery method may be added by selecting the option image in the dropdown menu.

An example scenario of the Fax module is shown below, where an incoming call is forwarded to the Sales Fax module.

image

Call Recorder

The Call Recorder module allows for phone calls to be recorded. The call contents are sent to a predefined destination immediately after the termination of each call. ( i.e email address, dropbox etc)

record calls module

Configuration options for the call record module are as follows

  • The name of the Call Recorder module.

  • The delivery method for the file containing the contents of the recorded call, with the options being Email, Dropbox, FTP, SFTP, Google Drive or OneDrive. If the required delivery method has not previously been configured by using the Delivery Methods option listed under the “Settings” menu, a delivery method may be added by selecting the image option in the dropdown menu.

A usage example for the Call Recorder module is shown below, where this module is inserted between a Phone Number and Ring Group module.

All incoming calls to Mike Brown are recorded, and the audio file is forwarded to a predefined destination immediately after the termination of each call.

Caller Router

The Caller Router module allows for incoming calls to be forwarded to different modules, depending on the originating phone number or CLI (Calling Line Identity). Users have the option of matching complete phone numbers, or forwarding calls based on the prefix.

caller router module

Caller routing is based on a simple “Allow/Disallow” logic, depending on the setting of the “behavior” indicator associated with each listed number/prefix. The Caller Router module must be connected to two child modules such as Ring Group or Voicemail modules, to which calls are forwarded according to this “Allow/Disallow” logic.

The Caller Router module includes both green (“Allow”) and red (“Disallow”) right-hand sockets for cable connections. A cable originating from the green socket corresponds to the “Allow” option, while a cable originating from the red socket corresponds to the “Disallow” option.

In the figure below, the Caller Router module is configured to include the phone number prefixes 1416,1437, and 1647, with the “behavior” set to “Allow” for all the listed prefixes. Therefore, if incoming calls are received with a CLI matching any of these prefixes, then those calls are routed via the green socket to the Toronto Voice Menu. All other incoming calls are sent to the Ontario Voice Menu.

The settings available for the “Caller Router” module are as follows

  • Name - The name of the Caller Router module.

  • Default Route - Specifies which route (“Allow” or “Disallow”) to which the call will be forwarded if the incoming phone number does not match any of the configured routing rules.

  • Algorithm - An algorithm to be used for number routing, with the options being “Prefix” or “Number”. “Prefix” matches are determined according to “number starts with” logic, while “Number” matches require an exact CLI match. Note that either the “Prefix” or “Number” option may be selected, and these algorithms may not be mixed on a single Caller Router module.

  • Numbers and Behavior - A list of phone number/s or prefix/es required for the caller routing logic, together with an “Allow/Disallow” indicator that defines the call routing behavior for that specific number or prefix. Alphanumeric inputs (both letters and numerals) are supported, with letters being case sensitive.

There are no limitations as far as the format (such as E.164) or the number of characters used in listing these phone numbers/prefixes, as to allow for maximum flexibility in any given CLI scenario. Multiple numbers or prefixes may be added as needed.

  • In the figure above, incoming calls with a CLI prefix of 1647 or 1416 will be forwarded to the module connected to the green (“Allow”) socket of the Caller Router module, while calls having a prefix of 1514 will be forwarded to the module connected to the red (“Disallow”) socket.

Phone numbers or prefixes previously added to the Caller Router module may be deleted by clicking on the red x icon to the right of the number.

Blocklist

Or also called a blacklist and/or spam filter

The Blocklist module is used to block incoming calls received from specific phone numbers and empty or alphabetical CLIs.

If the calling number matches the configured settings or phone number exactly, then the incoming call will be automatically terminated and will not be passed to other modules connected to the right-hand side of the Blocklist module.

blocklist module

Available settings for the blocklist module are as follows

  • The name of the Blocklist module.

  • Additional Settings:Block empty Caller IDs - When the source number field is empty or null, the call will be terminated.Block Caller IDs containing letters - When the source number field includes alphabetical characters, the call will be terminated.

  • The blocklisted phone number/s - A list of the exact phone numbers to be blocked. There are no limitations on the format (such as E.164) or the number of digits used in listing these phone numbers, so as to allow for maximum flexibility in CLI matching. Multiple numbers may be added as needed.

Phone numbers previously added to the Blocklist module may be deleted by clicking on the red x icon to the right of the number.

A simple usage example of the Blocklist module is shown below, where the blocklist filter is applied to all incoming calls before those calls are processed by the company’s voice menu.

Queue

Or in other words, the module used to manage callers “On-hold”.

The Queue module causes incoming calls to be placed in a queue before those calls are passed on to queue members (destinations). This allows a large number of calls to be handled, such as in a call center.

The Queue module includes a ring strategy that is used to define how the calls are divided between queue destinations. Music or other messages may be played to callers while they are waiting in the queue, and ringback tones are supported, allowing audio clips to be played to callers before their call is answered.

queue module

The Queue module includes multiple destinations to which calls are forwarded as per the selected ring strategy, and these destinations are added to the Queue module as “contacts”. The only cable exiting the Queue module is for a “queue timeout”, defining the call path if the queue wait time exceeds the maximum configured value.

In the example below, incoming calls are forwarded to the Queue module, which includes three queue destinations. If a call is not answered within the configured queue timeout, then that call is sent to the “Queue Timeout” Voicemail module.

Queue / On-Hold Configuration

  • The name of the Queue module. For example “On-hold for Acme Support”

  • The ring strategy to be used for this queue.

  • Ring strategy options:

    • Round robin – calls are allocated to members using a round robin policy, noting the last member who answered a call.

    • Ring all - ring all available queue members simultaneously until one answers.

    • Random – randomly ring a single queue member.

  • Additional ring strategies such as least recent and fewest calls will be implemented in future versions.

  • The queue timeout defines the maximum time that a call may remain in the queue without being answered, before that call is forwarded to a connected module according to the “queue timeout” logic. Valid values are from 00:05 (five seconds) to 15:00 (fifteen minutes). Note that if there is no call path defined for such a timeout event, then the call will be terminated on the occurrence of a “queue timeout”.

Contacts Tab - Queue

The Contacts tab is used to define the queue members or destinations. Each queue destination consists of a contact and an associated contact method, and these contacts may be directly configured from the Queue module or may be pre-configured by using the Contact Center. The section Contact Center provides full details on how to configure contacts and contact methods.

Selecting add queue allows the user to add a queue destination from a drop-down menu of pre-configured contacts and their associated contact methods. Alternatively, a new contact may be added by selecting add contact

Media Tab - Queue

The Media tab is used to set both “music-on-hold” and “ringback tone” playlists. This allows pre-recorded music, messages, commercials or any other audio clips to be played to callers before their call is answered (ringback tone), or while an active call is put on hold (music-on-hold).

The section Media Center provides full details on how to configure and manage media files. It should be noted that playlists must be pre-configured in the Media Center before adding modules to the workspace such as Queues , which may require these audio files.

CLI Rules Tab - Queue

The CLI Rules tab is used to override the Source Caller Name. This functionality allows flexible Caller Name configurations and may be used to differentiate SIP calls from the PBX (Phone system).

CLI Rules can help you identify which Queue module the call is coming from.

Note: By default, the CLI Rules feature is disabled.

Multiple Destinations - Queue

Selecting add queue allows the user to add a multiple ring destination from a drop-down menu of pre-configured contacts and their associated contact methods.

Time Router

The Time Router module allows for incoming calls to be forwarded to different modules, based on user-defined day and time intervals.

time router module

This module includes flexible time intervals, and call routing is based on a simple “Yes/No” logic. The Time Router module must be connected to two child modules such as Ring Group or Voicemail modules to which calls are forwarded according to this “Yes/No” logic.

A cable originating from the green socket on the right-hand side of the Time Router module corresponds to the “Yes” option, defining the routing of calls that are received within the configured day/time interval. A cable originating from the red socket corresponds to the “No” option, defining the routing of calls that are received outside of the configured day/time interval.

In the figure below, if incoming calls are received within the configured day/time parameters, then those calls will be forwarded to the sales queue. Otherwise, the calls will be forwarded to the Voicemail module where a pre-recorded “after hours” message is played to the caller, and the caller may leave voicemail.

time router example

The available settings for the time-router module are as follows

  • The name of the Time Router module.

  • The time zone that is applicable to the configured time routing intervals. Note that a “System” time zone option is available, designating the time zone for this Time Router module, to be the same as the time zone selected under General Settings.

  • The time routing intervals. By selecting add time interval, day/time intervals are created, defining the days of the week and the start/stop times for which incoming calls will be routed according to “Yes/No” logic. For each time interval, the day/days of the week to be included in this rule must be selected by clicking on that particular day, highlighting it in blue. Thereafter, “from” and “to” times must be configured for this interval. Times may be configured by using the simple time-selection menu provided, or may be entered manually in the form HH:MM using a 24-hour format.

create time interval

Each Time Router module may have multiple time intervals, allowing highly flexible options for forwarding calls. In the figure below, two separate time intervals have been configured, with incoming calls received from Monday to Friday from 08:00 to 17:00, as well as calls on Saturday from 09:00 to 13:00 being sent to the module connected to the cable labeled “Yes”.

Notification

The Notification module is used to provide alerts via email when a specified event occurs. For example, a notification may be issued when a caller joins a conference or when a queue timeout occur.

notify module

The available settings for the Notification module are as follows

  • The name of the Notification module.

  • The delivery method for the notification message, with the only option being Email. If the required email address has not previously been configured by using the Delivery Methods option listed under the “Settings” menu, a delivery method may be added by selecting the add delivery|directo296x47, 50% option in the dropdown menu.

An example scenario using the Notification module is shown below. Calls that arn’t answered by any of the members of the sales queue within the defined timeout period are passed on to the Notification module, and a notification email is sent.