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Setting Primary Phone Numbers
Configuring telegram as SIP softphone
Yealink T19
Yealink T21
Yealink T28
Snom D120
The "Extensions" section displays a list of internal user accounts for employees. On the left side of each employee, the status of the authorized device is displayed. If the device is successfully authorized under the respective internal user account, a green circle is shown; otherwise, it appears gray.
In the search bar, you can find the desired contact. You can search by the employee's name, internal number, mobile number, or email address.
The form also provides the ability to sort the list of employees by name, internal number, mobile number, or email address. There are buttons for copying the account password to the clipboard, editing the account, and deleting the account.
To add a new employee, simply click on the "Add new extension" button.
On the "Basic Parameters" tab, you can configure the general settings for an employee's internal account:
Username: This value will be used for name substitution and displayed in the corresponding field on the phone screen.
Internal Number: This is the employee's internal extension number, which is also used as the login when connecting the phone.
Mobile Number: It is used for additional routing purposes.
Email Address: It is used for email notifications.
Password for SIP
Please ensure that the passwords for SIP accounts meet the following requirements:
The password length should be greater than eight characters.
The password should contain both uppercase and lowercase letters. The password should include numbers and special characters: "-", "_", "[]", "{}", "@", ";".
By setting complex passwords, you can enhance the security of the user accounts and protect them from unauthorized access.
Accesses the Advanced drop-down list.
Redefining the set string
In the "Redefining the set string" field, enter the dialing rule for mobile numbers according to your provider's requirements.
If you want employees to have the ability to record conversations, you can enable the Сall recording feature.
The setting determines how DTMF (Dual Tone Multi-Frequency) signals are transmitted over the SIP protocol. DTMF signals are used, for example, when dialing phone numbers or interacting with IVR systems.
This setting allows you to specify the transport protocol used for this account. The transport protocol determines how data is transmitted over the network. The most common transport protocols used in SIP (Session Initiation Protocol) are UDP (User Datagram Protocol), TCP (Transmission Control Protocol), and TLS (Transport Layer Security)
The subnet described in the "Network Firewall" section specifies the allowed subnet for this account. It determines which IP addresses or networks are permitted to connect to this account. Connections originating from other subnets will result in authentication errors.
This field is used to modify/override the configuration files of Asterisk. You can override almost all parameters. For example, when using chan_pjsip, a SIP account for an employee is described by the following sections:
To override fields in the sections, you should fill in the "Additional Parameters" field as follows:
On this tab, you can set rules for call forwarding when the employee is unavailable, busy, or does not answer.
Set the time period in seconds during which the call will be directed to the employee's internal account. If the employee cannot answer the call within the specified time, indicate to which number the call should be forwarded. By default, the call will be redirected to the employee's mobile number.
You can also specify the numbers to which the call should be redirected in case of busy and unavailable status.
Feel free to configure these parameters according to your preferences and requirements.
An IVR menu includes options for routing incoming calls using an interactive voice menu. It allows callers to navigate through a series of menu prompts using their telephone keypad or voice input. The IVR menu typically provides various choices or options for callers to select based on their needs or preferences. Each option in the menu can be associated with a specific action or routing destination, such as transferring the call to a particular department, providing self-service options, or connecting the caller to a specific extension or queue. The IVR menu enhances the caller's experience by offering a self-service mechanism and streamlining call routing based on their selections.
Before creating an IVR menu, it is necessary to upload audio files that will be played to callers when they contact your company. The audio files can be added in the "Telephony" -> "Sound files"
Choose your music file using "Upload a new file"
Additionally, there is the option to record a file using a microphone if you connect to the MikoPBX over HTTPS.
Go to Telephony → IVR menu.
Press "Add New IVR Menu."
Set the name, number, and, if necessary, a comment for the IVR menu. Select the audio file that you uploaded in the previous step.
Configure Actions when you extend. In the first column, specify the extension number, and in the second column, set the addressing rule.
Set the number of retries before transferring to the default number.
Set the timeout for entering an extension number (value in seconds) after which the voice greeting will be repeated.
The Default extension is required in case the client does not enter an extension number (for example, due to technical limitations).
Enable the "Allow Dialing of any extension" toggle switch if needed.
Enter the IVR menu number that can be dialed to reach that specific IVR menu.
Press "Save."
The principle of operation of an IVR (Interactive Voice Response) is as follows:
When calling the IVR menu number, the Voice Greeting audio file starts playing.
During the playback of the voice menu, the caller can enter an extension number. The "Allow Dialing Any Internal Number" flag determines if callers can dial any internal number, including queues, IVRs, or internal extensions.
After the voice menu is played, there is a waiting period of the "Input Extension Timeout" for entering an extension number.
The total time allowed for entering the extension number is calculated as the sum of the audio file duration and the input extension timeout.
If the total time for entering the extension number expires, a repeat voice announcement occurs, and there is another waiting period within the timeout for the next IVR attempt.
If the user enters an incorrect number or does not enter any number at all, a repeat voice announcement occurs, and there is another waiting period within the timeout for the next IVR attempt.
The maximum number of attempts is set by the "Number of Retries" parameter before transferring to the default number.
Once the number of attempts exceeds the specified value, the call is redirected to the default number.
Queues allow you to:
Distribute phone calls among a group of employees (agents): You can create a call queue and add multiple employees to it. When a call comes in, the system automatically routes it to an available employee in the queue, ensuring a more even distribution of workload and increasing call handling efficiency.
Hold the customer on the line when all employees are busy: If all employees in the queue are occupied with other calls, the customer will be placed on hold until one of the employees becomes available. This helps avoid call abandonment and ensures better customer service.
Notify the customer of their position in the queue and approximate wait time: While the customer is in the queue, the system can provide information about their current position in the queue and an estimated wait time. This helps keep the customer informed and improves their waiting experience.
Display the queue name along with the customer's number on the employee's phone: When an employee answers a call from the queue, their phone displays not only the customer's number but also the name of the corresponding queue. This helps the employee handle calls more effectively and provide personalized service.
To configure call queues in MikoPBX, go to the "Telephony" section and select "Call Queue." Here, you can create and customize your queues according to your business requirements and customer service needs.
The default call duration for a queue is set to 300 seconds (5 minutes). After this time limit is reached, the call will be automatically terminated. To bypass this limitation, you can configure "Scenario 1" as described in the instructions for "Call Routing on Failures".
To add a new queue, perform the "Add a new call queue" action
In the queue creation form or dialog, you will find the following fields:
Queue Name: Enter a name for the queue. This name will be used for reference when setting up call routing rules.
Note: Provide a brief description or note about the queue. This information will be visible in the queue list, allowing you to provide additional details or instructions.
In the Queue Operators section, you can add an arbitrary number of employees (queue agents) and specify a call distribution strategy.
Here are the options for queue strategy:
Ring All: Calls are distributed to all available agents until someone answers the call (default behavior).
Least Recent: The call is routed to the agent who has been idle for the longest time within the queue.
Fewest Calls: The call is routed to the agent who has handled the fewest number of calls within the queue.
Random: A random available agent within the queue is selected to receive the call.
Round Robin: The call is distributed to each agent in a sequential manner, cycling through the list of agents.
Memory Hunt: The system remembers the last agent who answered a call and starts the distribution from that agent onwards.
When setting up a queue, you can choose one of these strategies to determine how calls are distributed among the agents in the queue. The strategy you select will depend on your specific call handling requirements and the desired distribution behavior
In this section, you can provide additional information:
Phone number for this queue - you can call the queue using this number from any internal employee extension. Calls can also be transferred to this number.
Short name for the queue - for display before the CallerID on the telephone device of the subscriber, for example, "consult."
Time attempt call to agents - the duration in seconds for which a call will ring on an individual agent's phone. After this time elapses, the call to the agent will be logged as a missed call in the call history. Once the ring time is over, the call will be routed to the next available agent based on the selected strategy.
The rest time of the agent after the processing of the call, before starting to accept new calls - the duration in seconds that is counted from the moment an agent finishes a call from the queue until they are ready to receive new calls. This period allows agents to update notes, complete necessary tasks, or take a short break before being assigned another call.
Receive New Calls During A Call - this toggle switch enables or disables the ability to receive new calls while the agent is already on a call. When enabled, agents can handle multiple calls simultaneously.
What the caller hears while waiting - During the wait for their call to be answered, the caller can hear either hold music or a ringing tone.
Background Music (MOH) - You can specify a unique audio file to be played to the caller during the wait, such as promotional materials.
Notify about current queue position - If all operators (queue agents) are occupied, enabling this toggle switch allows you to notify the caller about their position in the queue. If the Additional Audio Announcement option is activated, this announcement will supplement the information about the position.
Notify about estimated hold time - If all operators (queue agents) are occupied, enabling this toggle switch allows you to inform the caller about the approximate wait time for a call to be answered. If the Additional Audio Announcement option is activated, this announcement will supplement the information about the estimated wait time.
Additional notification - A sound message is played only if all participants in the queue are occupied.
Time in seconds to repeat all alerts periodically - Describes the interval at which to announce the queue position, wait time, and announcement.
The script #1 - In this scenario, you can configure the maximum allowable wait time for a client in the queue. If none of the queue agents can answer the client within the specified time, you can set a number to which the call will be redirected.
The script #2 - If there are no agents available in the queue (meaning no agents are currently logged into the phone system), you can specify a number to which the client's call will be transferred.
In these scenarios, as a redirection number, you can choose not only an internal extension but also options such as a conference, queue, IVR (Interactive Voice Response), or a special number within the dial plan application. These options provide flexibility in directing the call to different destinations based on your specific requirements or business needs.
The default call duration for the queue is set to 300 seconds (5 minutes). If you require a longer interval, you can specify a higher duration in Scenario 1 and provide a backup number to redirect the call to. This allows you to customize the wait time and ensure that if none of the queue agents can answer the call within the specified duration, it will be redirected to the designated backup number.
Supported file format mp3 and wav
Audio files in MikoPBX are used in various call scenarios and interactive voice menus (In IVR menu, during non-working hours, in call queues, for various system notifications, and in hold music.) to play voice greetings or notify customers.
The list of available sound files is displayed in the Telephony -> Sound files.
To add a new sound file, click "Add a new sound file".
Click "Upload a new file" and select a sound file.
Correct the file name if necessary.
Сохраните изменения.
When working over the https protocol, it is possible to record an audio file using a microphone.
Sound files are stored on the PBX along the path /storage/usbdisk1/mikopbx/media/custom
The function is available starting from version 2020.2.XXX
If a client gets into a queue during a call or is waiting for redirection, the PBX plays a melody for him. It is possible to download your own tunes for listening while waiting.
You can do this on the Music on Hold tab as described above.
Conference calling is used for conducting group discussions, meetings, or negotiations in cases where participants are unable to meet in person. It is also used when a particular matter needs to be discussed with multiple participants simultaneously.
The list of conference rooms is located in the Telephony -> Conferences section.
To create a new conference room, click the "Add conference".
You must specify the name of the conference and its internal number, by calling which you can enter this conference
To prevent unauthorized access to the conference by employees for whom the discussion is not intended, you can secure the conference room with a password. To do this, fill in the "Conference Pin" field. Only digits can be entered in this field, with a minimum requirement of at least one digit.
In this case, to join the conference, employees will need to enter the PIN code after dialing the conference PIN.
Communication is conducted solely through voice (no other means of information transmission besides speech are provided).
All participants can speak and hear each other simultaneously, ensuring duplex communication.
Participants use telephones (hardware or software) for communication.
Each participant dials the conference number. The first participant hears hold music until at least one more participant joins the conference. An employee can transfer their caller into the conference by using specific key combinations on their phone. Transfers can be made to both internal and external numbers. The key combination for transfers is set in the System -> General Settings -> Call Transfers section.
Example: An employee dials the combination **1111 (the combination for unconditional transfer), and their caller joins the conference as its first participant. The call is completed for the transferring employee, and to join the conference, they dial the conference number 1111.
The maximum number of conference participants is not limited.
Here you can find information about Call Detail Rescords (CDR)
Call History provides a log of all incoming, outgoing, and internal calls. It is located under Telephony -> Call History.
The Call History feature in MikoPBX enables users to:
Display all calls;
Filter calls based on criteria;
Visually identify missed calls from the call log;
Download or listen to call recordings.
Each entry in the call log contains information about:
The caller’s phone number (Who);
The recipient’s phone number (To Whom);
The date and time of the call (Call Date);
The duration of the call (Duration) – this excludes time spent on greetings or announcements.
Calls marked in red are missed calls. Their duration is logged as zero, and these calls cannot be played back:
For answered calls, users can listen to or download the recording. Call recordings are downloaded locally to your PC in .mp3 format.
Each call log entry provides detailed information about the participants involved.
To apply a filter, press Enter after entering the search criteria.
The search bar in the Call History page supports the following filters:
Phone Number Filter
You can search using either an internal staff number or an external client number.
Two Phone Numbers Filter
Enter two phone numbers separated by a space. For example, entering "74952293042 302" will display all answered calls between these numbers. Answered calls are those with a duration greater than 0 seconds, excluding greeting time.
Date Filter
When opening the Call History, the log defaults to the current date. To filter for a specific period, select the date range and click Apply.
