Call Flow

The basics

To start, a number is required. If you don't yet have one, press the Buy a number button (Fig. 1) to order a number from us. More on this process in the Phone number section. To see your available numbers, press the My numbers button (Fig. 1). By clicking on a number you will add it to the visual editor of Call Flow. Other numbers option is  Note, that if a number is grayed out, then it's already added to your Call Flow.

Fig. 1. (Phone Number section of TelTel web interface, Call flow subsection)

To define how incoming calls will be processed, elements must be used. By pressing the Elements button (Fig. 1) you will see a dropdown menu of all o

Each element in Call Flow has inputs and outputs. This is where the element is connected with another one.

For example, the Call to user element has an "in" point and four outputs.

For example, to say "when a call comes to my number, redirect it to a specific user", you would need to connect the phone number element with the Call to user element. To do that, we need to click on the output of the phone number and connect the arrow with the input of Call to user element.

Each element has unique settings that you can edit. To access them, double-click on the element. For example, this menu appears for the Call to user element:

* We will cover all the elements and their settings in the next section.

From this menu, we'll select the user to which the call will go.

Now, we can continue defining our logic using the outputs for the element. Please note, that it's not required to define logic for each output, you can leave some or all of them empty, depending on your needs.

• answer - define what happens if the assigned user has successfully picked up the call;
• no answer - define what happens if the assigned user didn't pick up the call;
• cancel - define what happens if the user cancels the call;
  
• error - define what happens if an error occurs.

At this point, we would like to say, "if the caller doesn't receive an answer, play an audio for him". Then, I would need to connect all the outputs for the unsuccessful pickup scenario with the input of a Play audio element.

This is a very basic example of a Call flow. You can add unlimited amount of elements and define a logic of any complexity that you need.

Call flow can quickly get messy, especially when defining a more complicated system. It's advisable to write descriptions for your elements, as well as keep the elements organized, so that your Call flow would be readable. A computer board esthetic is preferable. You can press on any point of a line to create an angle. 

Notice, that you can connect one output with multiple elements. This only works on outputs when that makes sense, for example the members output for the queue element. Don't connect a number's output or a Play audio output, for example, to multiple elements, as this can lead to errors.

To delete any element, just double-click on it and press delete. To mass delete, you can make a selection and click the cross.

You can also delete the Call Flow by pressing the red button Delete.

Don't forget to save your Call flow each time you make changes. Saving it will create a checkpoint, to which you can reverse in the Versions section. Even if you delete your Call flow, you can always restore it from here.

Now you are all set to start creating your own Call flow! Learn more about available elements in the next section.

Elements

In this section, we'll cover all existing elements and their settings.

You can click on any element below to learn more about it.

Call to phone

This element allows you to redirect the call to a selected phone. Note, that you can also select a CallerID that is verified in your account. It's also possible to select a dynamic parameter here, that can be passed as an output parameter from another element, such as the "HTTP and SMS notification".

Timeout specifies how long the system should dial the set number before dropping the call.

Call to SIP

This element allows you to redirect the call to a selected SIP device. You can check your SIP devices in the User section under SIP devices tab. It's also possible to select a dynamic parameter here, that can be passed as an output parameter from another element, such as the "HTTP and SMS notification".

If you need to forward the call to a SIP from a different provider, you can do this by selecting the Custom provider SIP. Enter the provider SIP in the format that is shown in the example.

Call to User

Similar to Call to phone and Call to SIP element, Call to user allows you to forward the call to an existing user in your account.

Voice mail

This element allows you to send Voice Mail.

Play audio

Play audio allows you to play any audio you insert for your client. It is advisable to use the .mp3 or .wav file extension. The audio files that you add here can be used for any other element that uses audio.

IVR - Voice menu

The interactive voice response element allows you to create a voice menu for your call flow. Through this element, you can define any kind of logic for any key that the client presses. If a key is not defined, it will be simply ignored by the system, if the client selects it. Create an audio file that explains the available options and the keys associated with them to your client. You can then add the audio through the audio element above and then select it in your IVR element menu.

IVR - number input

etc

Time routing

This element allows you to set up working hours for your call center. The client will be routed through true if he called during working hours, and through false if not. The working hours are applied in your local time.

Queue

The queue element is a very important element if you have a large volume of calls. This element will allow you to put your clients in queue, playing any kind of music or audio you select, and assign calls to your agents based on the settings you select. Connect the member's output with all of the or users or SIP's that should pick up the calls that came through the designated number. You can also define additional logic after a successful call through the finish output.

There are several strategies how the calls are assigned to your agents.

Ring all available members until one answers - this will send calls to all active assigned agents at once, until one of them doesn't pick up. This works best when you need to make sure that all calls are picked up as fast as possible, thus resulting in high ASR (answer rate).

Ring interface which was least recently called by this queue - this setting will look for the agent that has the highest waiting time on the line and assign the call to that one. This allows for an equal work load distribution. 

Ring the one with fewest completed calls from this queue - this strategy will calculate which agent has the least picked up calls count for the day and assign the call to that one. This would allow you to make sure all agents receive equal work load throughout their working time.

Ring random interface - this will simply call a random agent, without any kind of logic.

Round robin with memory - this will call each agent one by one until one answers. The system will remember where it left off and assign the next call to the next member in order, insuring an even work load distribution.

Rings interfaces in the order they are added - when you add the members that will pick up calls from this queue, the system remembers the order in which you add them. This strategy allows to call agents per this order.

There are also additional settings that you can toggle:

Timeout will define how long the system will allow the client to wait in the queue, before sending the call through the timeout output. If the timeout output is empty, the call will be dropped.

Member timeout defines how long the queue will wait for a designated user to pick up the call, until it passes it to the next one. This won't work if the "Ring all available members" strategy is selected, since it calls all members at once.

Join if empty defines if calls will go to the queue even if all assigned agents are offline. Recommended only if you have low offline time and the call center is working all the time. 

Ring in use defines if the calls will go to agents that are already on a call.

If no answer is an option to add additional logic for the scenario if the call has reached its timeout. This will force assign the call to a random agent from the list.

Random

This element has no parameters, and its only function is to send the call through a random route that is connected to its output. It can be useful, for example, to create variety for the music that plays on holds or in queue.

HTTP and SMS notification

Call state

This element will check if the current call is active and will send it through the according output. Useful when you need to be sure that the call is still active before sending the call to the rest of the call flow.

SMS notification

This element allows you to send a message to a selected number. You can add dynamic properties like (callerid) which is the client's number and (callto), which is the number the client called.

Email notification

Same thing as the SMS notifications, but this element will send an email.

Change CallerID

etc

Music on call, hold

etc

Caller name lookup DB

etc

Say price from parameter

etc

Routing by history

This elements checks if the current client on the call has already been in communication with one of the selected agents and will assign the call to that one. You can add any amount of SIPs or phone numbers to check. This allows your clients to communicate with already familiar agents.

Autodialer routing

etc

Text-to-speech

This element allows you to transform any text you need to a speech. We use the most advanced, incredibly realistic text-to-speech services available on the market - Microsoft Azure and Amazon AWS. Choose any voice you desire for a wide selection of options and languages. In the text you can also add dynamic parametrs from a HTTP nofication element. There is also a possibility to change the speed rate of the voice.

Short number

etc

Custom function

etc