Guaranteed dialing
Publication date: 02.06.2022
Guaranteed dialing module is available only in “CALLCENTER” tariff.
What it is?
Guaranteed dialing is a module of the UniTalk service that allows you to load employees with calls in automatic mode, so that employees do not need to do anything except pick up the phone and talk.
The essence of the guaranteed dialing functionality – we call the list of telephone numbers and when the handset is picked up, the call is immediately directed to the employees. Thus, employees simply pick up the phone or turn on auto-answer and be ready to speak when a call comes in.
Preparing to configure Guaranteed dialing.
Make sure that your project is in the CALLCENTER tariff, if not, contact UniTalk managers, they will be very happy, instantly and free of charge, to replace your tariff and will definitely offer to help with the setup, which will save you from further study of this manual.
Department сreation.
Now we will create a group of employees with whom we will need to connect clients when they pick up the phone.
1. Go to the UniTalk account section – IP Telephony – Departments.
Click the “Add Group” button.

2. Specify the name of the group. The name has no technical meaning, it is necessary to understand which lines of which employees are in which groups.
Examples of names: “Department No. 1”, “Alena’s Team”, “Operators 3rd floor”, “Interns”, “HARD department”.

3. The default will be “Internal line group (SIP)”, just do not change.
4. Enter a shortcut number. This is not required, but it allows you to dial this number on any SIP line and call all the lines in this group at the same time.
5. Click “Add” next to the Lines in the group block as many times as there are employees in the department.
6. In the appeared select the lines of all employees that should receive dial-in calls.

Done, the department is created.
Preparing an outgoing script.
If you plan to make a call from a number that is used for regular calls, then you do not need to do anything, otherwise follow the instructions further.
Go to the section of the UniTalk account – IP Telephony – Outgoing Scripts and click the button at the top of the “Add Script” page.

In the interface that appears, in the general rule, click “Specify” and remove the setting to use UniTalk numbers for calls.

Give a name that makes sense to you, for example “Dialing” or “Dialing script”. Select the number from which you want to dial and save the script.

Great, the outgoing script is ready.
Event processing

To call events and call end events you can bind the execution of actions. An action can be:
- sending a webhook
- sending a message to Telegram
- send an SMS
- perform an action in CRM
More detailed configuration of actions on the “Event processing” page. There are also instructions for setting up various actions.
Preparing and adding a list of numbers of people to be called.
Go to the section of the UniTalk account – Guaranteed dialing click the “Add” button.

There are 2 options, one for when you need to download a lot of numbers at once, the other for regularly downloading a small quantity of numbers, which is easy to select, copy and paste.
Option 1. Download table with numbers.
- Download file format.
- Supported extensions – .xls or .xlsx
- Quantity of numbers with no more than 25000.
- Column names phone, name, note.
- The “phone” column is the only required column.
- Numbers must be in international format (+380… or 380…). Spaces and non-numeric characters are removed automatically.
- The phone number for one call is unique, that is, duplicate numbers will not be added.
- Column “name” – a maximum of 45 characters, if more – the extra ones are cut off.
- Column “note” – a maximum of 200 characters, if more – the extra ones are cut off.
- The columns can be placed in any order.
- The presence of extraneous fields is irrelevant.
Example in the image below.
You can download the table by clicking on the gray area with the inscription “No file selected.”

Done, contacts loaded.
Option 2.
In cases where you need to regularly upload a small number of contacts, it is easier and more convenient to simply select the list of numbers and paste it into the list of numbers area.

Done, contacts loaded.
Number priority
Relevant for re-adding numbers.
Contacts can be added with or without priority. If you turn on the “High priority” setting and then download a list of numbers, all numbers from this list will be priority and will be called in the first priority. If you add numbers not including the “High Priority” setting, they will be added to the end of the list.
Operators
Select from the list the previously created department of employees who should receive calls by guaranteed dialing.

Calling mode

“By the number of streams” – call while there are free streams.
One steam is one simultaneous call. The number of streams should not exceed the number of channels of the number from which dialing is made.
Important!!!
It is necessary to leave several free channels of the number so that those people who will call back will get through.
If the channels of the number are loaded at 100% (dialing in 10 streams works from the 10th channel number), the person who calls, the call will be dropped and so it will not get into the UniTalk call history. We will be able to provide information on the number of such calls, but without the phone numbers of those who called.
This type of call is suitable when it is not particularly important to load all employees and/or quantity of calling numbers in relation to the quantity of channels of the number allows you to call everyone in the shortest possible time.
“According to free operators” – we call when there is or a free operator appears.
Only take into account:
operators on SIP lines,
groups of operators on SIP lines,
operator queues on SIP lines.
If all operators are offline, then calling is automatically paused.
“Smart selection” – works on the basis of the “free operators” mode. If there is a free operator – calling.
When there are no available operators, the algorithm can decide to start a call, based on the data from the completed calls of this calling, and it will start working after a sufficient number of successful calls have been made. This is the key difference from the previous “by free operators” mode.
This mode is suitable if the main task is to load all employees with calls with a minimum call waiting time.
Not suitable if after each call the employee needs time for some action.
Schedule
By default, a standard, as for Kyiv, working week is selected. If your clients are legal entities and plan to make phone calls during working hours, then “it’s ok the way it is!”.

The best conversion is achieved in the afternoon, people are a little kinder when they have eaten.
It’s better to exclude dialing:
while people are driving from/to work,
friday night,
in the mornings on weekends and after holidays, since during these time periods there will be a low probability of picking up the phone.
Recalling
In dialing there are always unsuccessful calls, some people do not pick up the phone or for other reasons the call will not take place. All such calls can be re-dialled. You can specify number of callbacks interval between them, callbacks with or without priority and select call statuses by separating those who do not need to call back, for example, those who picked up the phone and dropped the call (status – Answered (reset)).

In the case when dialing goes on a “cold base”, we do not recommend specifying a large number of repeated calls and a low minimum interval. People, seeing a large number of missed calls, may regard this as spam and blacklist your number or assume that something very important / urgent has happened and get nervous.
Settings
Depending on the dialing mode, the settings are different, except for “Name”, “Audio instead of beeps”, “Outgoing script”, “Speech synthesis settings”, “On audio error”, “Prefix for SIP”, “Send to CRM”, “Show name in SIP client” and “Show note in SIP client” – these settings are common for all modes.

Audio instead of beeps. The audio that the customer will hear while dialing employees after the person answers the dialing call. If the audio file is not needed and the call should be immediately directed to employees, then select “no audio”. You can upload your audio recording in the “Audio” section, the “Waiting on the line and queuing melodies” category. If specified, “beep melodies” that are specified in the incoming script will be ignored.
Outgoing script. Select the outgoing script in which the number from which the call should be made is selected.
Speech synthesis settings. This setting is used to select the speech synthesis settings profile that will be used for audio synthesis. You can add calls with text to be voiced from an xls/xlsx file or via the API. For more details about speech synthesis, follow the link.
On audio error. This setting allows you to choose what to do if an error occurs due to which the synthesized audio recording cannot be played. If during the call an audio file search error or an audio synthesis error occurred for the call, then either the general call audio will be used (the one selected in the “Audio instead of beeps” setting), or the call will end with the “Audio error” status. You can add calls indicating audio recordings or text for voicing from an xls/xlsx file or via the API.
Prefix for SIP. Helps employees to distinguish between a dialing call and a regular incoming call. Can be filled, for example “GD” and the employee will see the customer number in the format “GD-380501234567” instead of “380501234567”.
What calls to send to CRM.
“None” – calls will not be sent to CRM systems.
“Answered by the subscriber” – only calls answered by the subscriber will be sent to CRM systems.
“Answered by the subscriber and the operator” – only calls answered by the subscriber and the operator will be sent to the CRM system.
“All” – all calls will be sent to the CRM system.
Show name in SIP client. If enabled, then when dialing the operator in the SIP client, the caller’s name will be displayed as “phone-name”. If enabled and “show note”, it will be displayed as “phone-name-note”. You can add customer numbers with a name and a note in the “Numbers” section by uploading an .xls or .xlsx file and clicking the “Add customers” button.
Show note in SIP client. When enabled, the employee will see the customer number in the format “380501234567-Name-note” instead of “380501234567”. You can add customer numbers with a name and a note in the “Numbers” section by uploading an .xls or .xlsx file and clicking the “Add customers” button.
The following settings vary depending on the dialing mode.
“By number of steams” mode

Streams. The number of streams must be less than the number of channels of the phone number that is specified in the linked outgoing script. For those people who will call back on a missed call, a free number channel is needed so that they get through, the call displayed in the UniTalk call history and directed to employees if there are free ones.
If the channel is not enough, the call will be dropped during the call back, the person will not hear anything, and this call will not appear in the UniTalk call history, since the call will be canceled on the side of the telecom operator and will not reach UniTalk.
By changing the number of streams, you control the load on managers. Ideally, allocate a supervisory employee who will monitor the load and change the number of steams “on the fly“ to avoid both forced downtime and employee overload during the dialing process.
Overload situation. When some managers’ communication with the client is dragged on and they seem to fall out of the general call processing, calls to other managers begin to arrive more and more often.
Downtime situation. When in the list of numbers, there are a lot of invalid numbers in a row or they simply don’t pick up the phone because the working day is over and many are in transport.
Calling interval. Pause between making calls, starting a call, not ending. Only works when the number of steams is 1. Measured in seconds.
Call interval for answered calls only. The pause between calls will work only after the calls answered by the operator.
“According to free operators” mode

This mode has only one additional setting – this is the ability to:
Reserve extension lines only for call of dialling– when a call starts, one internal line is reserved and the reserve is removed only after the successful completion of this call.
Mark SIP line busy after a successful call for (sec) – a short pause for the operator after a conversation with the client, during which the line is marked “Busy” and a new call will not be received.
Mark SIP line busy if the conversation lasted at least (sec) – if the conversation lasted longer than the specified time, then the status of the SIP line will change to “Busy”
Pause the user if he does not receive calls. If the specified number calls of dialing, redirected to one of the internal lines ends in a row with the status “Operators did not answer the call” – the user assigned to this line will be put on pause. Pause – this feature is available in advanced user functionality;
Number of calls in a row before pause. The number of calls in a row with the status “Operators did not accept the call” after which the user will be paused;
“Smart Pick” Mode

Pause the user if he does not receive calls. If the specified number calls of dialing redirected to one of the internal lines ends in a row with the status “Operators did not answer the call” – the user assigned to this line will be put on pause. Pause – this feature is available in advanced user functionality;
Number of calls in a row before pause. The number of calls in a row with the status “Operators did not accept the call” after which the user will be paused;
Cold base. If the percentage of successful dialing to subscribers is small, thanks to this setting, you can start several calls at once. This will help reduce call waiting time for operators. Let’s say the percentage of dialing in the last 10 minutes is 20% and the cold base multiplier is 1, in this case, when it comes time to call, 5 calls will start at once;
Cold base multiplier. The higher the multiplier, the more calls will be launched, the lower the multiplier – the fewer calls.
Call speed regulator. Allows you to speed up or slow down the call. Less is slower, more is faster. 0 seconds – does not change speed.
It is not recommended to set the speed to more than 0 seconds, this will lead to more calls that will be made too early, before the operators are free from current calls and the client does not get a free operator.
Launch
After selecting all the settings, it remains only to save.

Done, dialing will start automatically. Calls will start arriving at employees after a short pause or at the next time according to the schedule.