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Description of the first login to the MikoPBX web-interface
Go to the MikoPBX installation console, remember the IP address that your PBX received.
Enter the received MikoPBX IP address in the web browser. The authorization page will be displayed. Log in using the default credentials:
After successful authorization, MikoPBX will automatically open the settings for changing the password:
After changing the password, the system will be fully operational. It is recommended to immediately configure the firewall rules. You can read about how to do this by following .
Use the following default credentials for the first login to the MikoPBX web interface:
Username: admin
Password: admin



This guide provides detailed steps to get started with MikoPBX and helps you quickly configure the system.
Follow the step-by-step instructions in the order presented for a quick and successful system setup.
MikoPBX is a full-fledged operating system for your hardware; it is not a standalone application. It is provided as an image file (*.iso, *.img, *.raw).
It supports various installation methods:
Installation on a standalone computer.
Installation in a virtual machine.
Installation .
Installation in a .
Follow the link for your preferred installation method and proceed according to the provided instructions.
After installation, you need to access the MikoPBX web interface for further system configuration. To do this, find the PBX's IP address in the MikoPBX console:
In this example, the IP address is 192.168.0.203. To access the web interface, enter this IP address into your browser's address bar:
After the first login, the system will prompt you to change your password.
For stable PBX operation, you need to configure the network through the Network and Firewall → Network Interface section. Detailed instructions for these settings can be found .
In MikoPBX, all local subnets can be defined in the Network and Firewall → Firewall section. The firewall is intended to restrict access to the PBX based on traffic type and subnets. Follow the setup instructions .
Fail2Ban blocks IP addresses exhibiting unusual activity; it can reduce the rate of failed authentication attempts and helps protect your PBX from hacking. Instructions to help with the setup can be found .
After completing the initial PBX setup, you can proceed to create accounts for your employees. This will assist you.
After adding employees, you need to connect providers to your PBX. Instructions for this section can be found . Instructions with examples of configuring real providers can be found .
At this stage, you need to set routing rules for incoming and outgoing calls: how calls passing through a specific provider will be handled:
To create routing rules, you may also need the following features:
The Marketplace allows you to extend the system's standard functionality using modules:
You can read more about Modules in MikoPBX in .
Information on registering in the MikoPBX Marketplace can be found .
Discription of system requirements for the MikoPBX system
An example of calculating the required channel bandwidth for different codecs for 30 simultaneous calls. PBX supports the most popular codecs:
G.711 - 4.67 Mbps
GSM - 1.68 Mbps
Description of MikoPBX and the sections you can find in the documentation. Introduction to the documentation.
Welcome to the MikoPBX documentation resource! Here you can find step-by-step instructions related to interacting with the MikoPBX PBX system. For your convenience, the documentation is organized into sections - just like in the web interface, making it very easy to navigate.
Thank you for choosing MikoPBX! ❤️
MikoPBX is a free telephony server with its own operating system and a simple, user-friendly web interface. It works with virtually any telephony technology in the world.
MikoPBX is a fully modular interface for Asterisk, written in PHP and JavaScript. This means that you can implement absolutely any additional Asterisk telephony functionality within MikoPBX. Moreover, if you develop a useful module, you can place it in the public repository and make it available to all MikoPBX users. Additionally, MikoPBX has very low hardware requirements:
In this article you will find instructions on how to install MikoPBX on a separate computer.
MikoPBX supports installation on a standalone computer. There are two installation methods:
Live USB — writing the image to a flash drive and then installing it onto another disk.
Bootable USB — installing the system directly onto the USB drive itself, allowing you to boot and run the system from it.
This section provides image writing instructions for all supported operating systems (Windows, MacOS, Linux).
If the logs do not provide a username and password, use the default credentials:
Username: admin
Password: admin


In this article you will find instructions on how to install MikoPBX using various cloud services.
MikoPBX supports installation via many cloud services. In this section you can find detailed instructions for them. Click on an item in the list below to go to the instruction for a specific virtual machine:
AWSMicrosoft AzureGoogle CloudHetzner cloud (In dev)Digital OceanAlibaba CloudVultrIn this article you will find instructions on how to install MikoPBX using various virtual machines.
MikoPBX supports installation using many virtual machines. In this section you can find detailed instructions for them. Click on an item in the list below to go to the instruction for a specific virtual machine:
VMware ESXiVMware FusionVirtualBOXVMware Workstation ProHyper-VProxmoxUTMG.729 - 1.38 Mbps
800 Mb hard disk for the main system
A 50+ Gb hard drive for recording conversations
1 (2 cores) x86-64 processor
2 GB of RAM
Network Adapter
We recommend using two hard drives for PBX deployment.
A PC with such parameters, in our tests, holds 38 simultaneous incoming calls under the conditions:
10 agents are connected to the queue (all online)
Every second a new call comes in
Approximately, 1 hour of conversation takes up 14MB of disk space. The recommended size for the disk storing call recordings is at least 50 gigabytes.
5 - 10
1 GHz x86-64, 512 MB RAM
Up to 25
3 GHz x86-64, 1 GB RAM
Over 25
2 CPUs 3 GHz x86-64, 2 GB RAM or more
To get started, you should install MikoPBX using any method convenient for you. Below are installation options. By clicking on their names, you can access detailed articles:
Installation on a standalone computer.
Installation in a virtual machine.
Installation using cloud services.
Installation in a Docker container.
After installation, you can begin exploring your PBX system. The "User Guide" documentation will help you with this, providing detailed information about specific sections:
For additional help with getting started quickly, you can refer to this article.
If you have familiarized yourself with the basic setup and operation of MikoPBX, you can expand its functionality using modules.
Modules allow you to add extra features to your system. You can learn more about them step by step:
Registration in the MikoPBX Marketplace – here you'll find a detailed description of the registration process and its specifics.
Module Management – this section provides a detailed explanation of how to install and manage modules.
Documentation on Specific Modules – in this section, you'll find detailed descriptions of each module, as well as steps for configuring and using them.
In this section, you can find answers to your questions and solutions that will help you expand the functionality of basic features. This section, like the main documentation, is divided into categories for easy navigation.
If you have a question that isn't covered here, you can seek assistance in the Telegram Community, where MikoPBX users help each other resolve issues and needs related to the PBX system.

MikoPBX Installation Options Using AWS
MikoPBX Installation Guide using Google Cloud
Installing MikoPBX in Google Cloud can be done in two ways: using an image from the Google Cloud Marketplace or from an image based on a file uploaded from the MikoPBX distribution. The first method provides quick and easy deployment of the standard version of MikoPBX, while the second is suitable for intermediate releases.
MikoPBX Installation Guide using Docker container
MikoPBX can be run in Docker using two main methods. The first method involves running the container directly using a Docker command with the necessary parameters. The second method involves using Docker Compose, which simplifies managing multi-container applications and allows the entire configuration to be described in a yaml file, making the deployment and maintenance of the system more convenient.
How to view and filter call history in MikoPBX
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;
Description of the MikoPBX telephony section
The "Telephony" chapter in the MikoPBX documentation contains detailed information and instructions related to setting up and using telephony in the system.
Extensions in MikoPBX are individual users of the system who are assigned internal numbers for making and receiving calls. They have personal accounts that allow you to configure access rights, call forwarding and other personal settings in the system.
In this article, you will find detailed documentation on adding new employees to the station, setting up their rights and profiles. In addition, information about their additional parameters.
Call queues in MikoPBX are a feature that allows you to distribute incoming calls between a group of operators, holding calls in a queue until an operator becomes available. This ensures efficient management of a large call flow and improves customer service.
In this article, you will find detailed documentation on creating and configuring such queues.
Preparation guide for MikoPBX using Docker
Before working with Docker, you need to install Docker and Docker Compose themselves. Here's how to do it:
Before creating the container on the host machine, it's necessary to create a user and group with limited permissions, as well as a folder for storing configuration settings and call recordings
Command to connect to the PBX console:
Command to connect to the PBX console menu:
Connecting to sngrep for SIP analysis
Description of the Modules section in MikoPBX
The "Modules" section in MikoPBX is an interface for managing additional functional components of the system, which includes two subsections: "Module Management" and "Dialplan Applications".
Managing these subsections allows you to configure MikoPBX as flexibly and efficiently as possible, expanding the functionality of the telephone system and adapting it to the unique requirements of the organization.
Registration in the MikoPBX Marketplace does not affect the basic functionality of the system. You can use MikoPBX to work with calls without registration and installation of additional modules. However, we recommend that you go through the registration procedure in the marketplace to get the opportunity to expand the functionality of the system.
Registration will give you access to additional modules and extensions.
Module management in MikoPBX is an interface for managing additional system components that expand its functionality. Here, administrators can install, update, enable or disable modules, adding new features or integrations with external services. This section allows you to adapt the system to the specific needs of the company, ensuring flexibility and scalability of the telephone network.
Description of the "System" section in MikoPBX
The "System" section in MikoPBX is the interface for managing general settings and parameters of the telephone system. Here, administrators can configure core system parameters, manage updates, date, and other functions that ensure stable and secure operation of MikoPBX. This section allows you to control and optimize the system's operation at the entire infrastructure level.
In the "General settings" section of MikoPBX, administrators can manage the main system parameters, such as call recording retention settings, notifications, log parameters, voice prompt language, and many other system options. This section provides control over general functions and behavior of MikoPBX, allowing you to optimize the system's operation according to the organization's needs.
The "Time Settings" section in MikoPBX is an interface for configuring system date and time parameters. Here, administrators can set the current date and time, choose a time zone, and configure synchronization with Network Time Protocol (NTP) servers. Correct date and time settings are important for accurate event logging, call logs, and the operation of schedule-dependent functions, ensuring system synchronization with other network devices and services.
The "Mail settings"
Description of the Network and Firewall section in MikoPBX
The "Network and Firewall" section of MikoPBX is an interface for configuring network settings and managing the system's firewall. Here, administrators can configure IP addresses, network interfaces, and create firewall rules to protect the system from unauthorized access. This section ensures the secure and stable operation of MikoPBX in the organization's network infrastructure.
The "Network Interface" section in MikoPBX is an interface for configuring the system's network connection parameters. Here, administrators can manage IP addresses, subnet masks, gateways, and other network settings for each network interface. This allows MikoPBX to be correctly integrated into the organization's network and ensure its stable operation in accordance with the requirements of the network infrastructure.
The "Firewall" section of MikoPBX is an interface for configuring the system's firewall. Here, administrators can create and manage network traffic filtering rules, controlling access to MikoPBX and protecting it from unauthorized access and network threats. Configuring the firewall ensures the security of the telephone system, preventing potential attacks and ensuring stable operation in the organization's network infrastructure.
The "
In this subsection, you can configure the clock and calendar settings.
You need to set the time zone correctly to ensure accurate system time display. If the time zone is not set, notifications and call history will be recorded with incorrect timestamps.
To configure the time zone, go to the "System" section and select "Time Settings":
To manually set the time, toggle the switch "Adjust the time manually."
A method to transfer MikoPBX to another host using backup
This method involves creating a backup of your current MikoPBX configuration, transferring it, and restoring it on the new server. It’s simple to implement and suitable for small systems. This approach is convenient for users with minimal technical experience.
First, create a backup of your previous system. You can find detailed instructions in .
Select the data you want to transfer and wait for the process to complete.
Description of the Maintenance section in MikoPBX
The Maintenance section of MikoPBX is an interface for managing the technical aspects of the system and ensuring its stable operation. Here, administrators can perform tasks on data backup and recovery, software updates, system status monitoring, and event log management. This section helps maintain the functionality of the telephone system, promptly detect and eliminate possible problems.
The "PBX update" section in MikoPBX is an interface for managing system software updates. Here, administrators can check for new versions, install updates, and view the change history. Regular use of this section ensures that MikoPBX is up-to-date, secure, and stable.
The "System log entries" section in MikoPBX is a tool for monitoring and analyzing the status of the telephone system. Here, administrators can view event logs, check the status of various services and system components, and test connections and calls. Using this section helps to promptly detect and eliminate technical problems, ensuring stable and efficient operation of MikoPBX.
The "Reboot" section in MikoPBX is an interface for securely managing the state of the telephone system via the web interface. Here, administrators can reboot the system to apply new settings or shut it down gracefully for maintenance. Using this section prevents possible errors and ensures stable operation of MikoPBX. In addition, the possibility of rebooting via the console will be discussed.
When a new is added to the PBX, a SIP account with a numeric internal number is created on the PBX. In some cases, for security reasons, it is necessary to change the name for authorization of this employee.
When configuring SIP Clients, you can often see two key parameters:
Username - usually equal to the account ID, in the case of MikoPBX equal to the internal number
Auth name - username for authorization. In the case of MikoPBX is equal to the internal number
To change the authorization name:
Music (MOH) is played to the client while waiting
Modules on the PBX is not installed
The "Host system" must run on Linux 5+. Tested on Debian 11, Ubuntu 21.04, and Ubuntu Server 22.04 LTS.




Installation by writing the image to a USB drive (Live USB)
Installing the system directly onto a USB drive (Bootable USB)


AWS deployment guide
Detailed instruction on how to create an AMI image
AWS Marketplace
Instructions for installing a ready-made image from AWS Marketplace
AWS Terraform script
Installation instructions using Terraform (IaC)


Google Cloud Marketplace
Instructions for installing a ready-made image from Google Cloud Marketplace
Google Cloud deployment guide
Detailed instructions for uploading MikoPBX from an image based on a .vhd file from the distribution


In this article, you will find documentation on creating and configuring an IVR menu.
Conferences in MikoPBX are a feature that allows you to organize group phone calls with multiple participants at the same time. It allows you to hold group discussions, meetings and appointments over the phone, improving communication both within the company and with external partners.
In this article, you will find documentation on creating and configuring conference rooms.
Sound files in MikoPBX are audio recordings that are used by the system to play various messages, such as greetings, announcements, IVR menu instructions or waiting signals. They allow you to personalize the audio content that callers hear, improving interaction with the system and providing the necessary information.
In this article, you will find detailed information about them, as well as how to add and edit them.
Call detail records in MikoPBX is a log that stores information about all incoming and outgoing calls through the system. It provides detailed data about each call, including time, duration, participant numbers and status, which allows you to analyze communications and optimize the operation of the company's telephone network. In this article, you will find information about storing call records and their filters.

MikoPBX dialplan applications are a set of tools that allow you to set up individual call processing scenarios within the system. With their help, you can define a sequence of actions that the system will perform when a call is received or made. This may include redirecting a call to a specific extension, playing special audio messages, requesting additional information from the caller, or performing other functions.
Using dialplan applications, you can flexibly customize the logic of the telephone system to the needs of your business without delving into complex programming. This makes it easier to create complex call processing scenarios, allowing you to improve the efficiency of communications and improve the level of customer service.

The "Asterisk Manager Inteface (AMI)" section in MikoPBX is an interface for configuring access to the Asterisk Manager Interface (AMI). Here, administrators can manage AMI connection parameters, such as enabling or disabling access, specifying login credentials for authentication. Configuring AMI access allows external applications or scripts to interact with the MikoPBX system for monitoring and managing calls, expanding the telephone system's functional capabilities.
The "System files customization" section in MikoPBX provides administrators with the ability to directly modify or supplement the system's standard configuration files. Here, you can make individual settings that are not available through the standard web interface and adapt the system's behavior to the specific requirements of your organization.
This section is intended for advanced users who have a deep understanding of the structure and operation of MikoPBX. With its help, you can:
Edit configuration files: Make changes to existing files or add new parameters.
Override standard settings: Change default values for certain functions or modules.
Add your own scripts or modules: Expand the system's functionality by integrating custom solutions.
It is important to note that incorrect modification of system files can lead to unstable operation or system failures. Therefore, it is recommended to create backups before making changes and to carefully check the correctness of the settings.

sudo docker exec -it mikopbx sh# Update package list and install required dependencies
sudo apt-get update
sudo apt-get install -y ca-certificates curl
# Add the GPG key for Docker's official repository
sudo install -m 0755 -d /etc/apt/keyrings
sudo curl -fsSL https://download.docker.com/linux/ubuntu/gpg -o /etc/apt/keyrings/docker.asc
sudo chmod a+r /etc/apt/keyrings/docker.asc
# Add Docker's repository to the APT sources list
echo \
"deb [arch=$(dpkg --print-architecture) signed-by=/etc/apt/keyrings/docker.asc] https://download.docker.com/linux/ubuntu \
$(. /etc/os-release && echo "${UBUNTU_CODENAME:-$VERSION_CODENAME}") stable" | \
sudo tee /etc/apt/sources.list.d/docker.list > /dev/null
# Install Docker CE and Docker Compose
sudo apt-get update
sudo apt-get install -y docker-ce docker-ce-cli containerd.io docker-buildx-plugin docker-compose-plugin
# Verify Docker Compose version
sudo docker compose version# Creating a new user (e.g., www-user) without superuser rights
sudo adduser --disabled-password --gecos "" www-user
# Creating directories for data storage
sudo mkdir -p /var/spool/mikopbx/cf
sudo mkdir -p /var/spool/mikopbx/storage
# Granting the created user ownership of the directories
sudo chown -R www-user:www-user /var/spool/mikopbx/sudo docker exec -it mikopbx /etc/rc/console_menusudo docker exec -it mikopbx sngrepVisually 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. When downloading a recording, you can choose WebM (Opus), MP3, WAV, or OGG (Opus) format.
Each call log entry provides detailed information about the participants involved.
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.
Specific Field Filter
You can add a prefix to search only in a specific field:
src:74952293042 - search by caller number;
dst:302 - search by destination number;
did:74952293042 - search by DID number;
linkedid:mikopbx-... - search by the unique call identifier.
If no prefix is specified, MikoPBX performs a general search by caller number, destination number, DID, and employee name.
Date Filter
When opening the Call History, MikoPBX selects a date range based on the latest call records. To filter for a specific period, select the date range and click Apply.

Download your archive by clicking the corresponding button in the "Backup Module" section:
On the new host (server) with your MikoPBX installation, restore from the archive by clicking "Upload backup file":
After this, your system will be restored from the archive. This method is ideal for transferring small amounts of data.


Open the employee's account for editing
Go to "Advanced Settings" → "Advanced SIP Settings"
Add the following lines:
Instead of MIKO204, specify a name for authorization, it is advisable to use both letters and numbers
[auth]
username = MIKO204Description and configuration of incoming routing
In this section, you need to create rules and templates for distributing incoming calls for providers created in MikoPBX. The rules for incoming calls describe the route of a call from the moment it arrives at the PBX to the moment it is completed. You can create an unlimited number of inbound routing rules. You can create several rules for one provider.
Rules are listed in order of priority. If no one answers the incoming call within the time interval specified in the rule, the call will be routed to the next priority rule. Rules can be moved up and down in the list, that is, their priority can be changed by dragging them by the arrows.
If the call is not answered according to any of the rules, the Default incoming route is used.
The following actions are available and can be specified as the default rule:
Play busy signal - the client will play a busy signal and the incoming call will be ended;
Hang up;
Redirect the call - the call can be transferred to a number that you can select in the field located to the right of the action. You can select an IVR menu, call queue, conference, or employee extension number as the number for transfer.
For one provider, you can describe several incoming routes.
First, the call goes along the upper route. If the client does not get through, then the call goes according to the lower rule (lower priority). If the client does not get through via the second route, then the call goes through the default route.
To add a new incoming routing rule, click the Add a new rule button.
In the Note field, describe the route you want to implement. In the future, this will help you debug the call circuit.
Select the Provider for which you are creating a new incoming call distribution template.
The additional DID number is the number the client called you on. This field is optional and should be completed if you need to route calls more accurately.
At the next step, you need to indicate to which phone number the incoming call from the client will be sent. The telephone number can be IVR menu numbers, call queues, conferences, or employee internal numbers.
Specify the time during which the call will be sent to the phone number you specified.
If after the specified time interval no one answers the incoming call, the call will be routed to the next priority rule.
Description of the MikoPBX routing section
The "Call Routing" section in MikoPBX is an interface for configuring call direction rules within the telephone system. Here, administrators can determine how to handle incoming and outgoing calls by setting conditions and routes for efficient distribution of calls among employees, departments, or external lines.
Telephony providers in MikoPBX is the system section where connections to external communication operators are configured via internet protocols for IP telephony. Here, administrators can add and configure SIP trunk accounts or other types of connections that allow the system to make and receive calls to and from landline and mobile numbers.
In this article, you will find detailed documentation on connecting providers to the system, their configuration, and features.
Incoming Routing in MikoPBX are a set of rules that define how the system handles incoming calls from external telephony providers. With them, administrators can set call directions based on various conditions such as the caller's number, time of day, or the specific number the call was received on. This enables automatic distribution of incoming calls to specific employees, departments, IVR menus, or call queues. Configuring incoming routes helps optimize call handling and improve customer service quality by providing flexible and efficient management of the company's telephone communications.
In this article, you will find detailed documentation on configuring incoming routing.
Outgoing Routes in MikoPBX are a set of rules and settings that determine how the system handles outgoing calls from employees to external numbers. With them, administrators can manage call direction through various telephony providers or communication lines based on certain conditions such as the dialed number, prefixes, time of day, or user access rights. This helps optimize communication costs, distribute load between channels, and implement security policies by restricting or allowing certain types of calls. Configuring outgoing routes provides flexibility and control over outgoing telephone communication, contributing to the effective operation of the company's communication system.
In this article, you will find detailed documentation on configuring outgoing routing.
Off-Work Time in MikoPBX is a tool for setting up call handling rules during periods when the company is not operating, such as at night, on weekends, or on holidays. With it, administrators can define how the system will handle incoming calls during off-hours: redirect to voicemail, play special voice messages, or forward calls to the mobile numbers of on-call staff. This allows for proper interaction with clients outside of working hours and maintains a high level of service.
In this article, you will find detailed documentation on setting up off-work time for your system.
Mail setup for the proton.me service
First, go to your Proton account settings (link).
Then go to the "Proton Mail" -> "IMAP/SMTP" section.
Scroll down to the "SMTP submission" section. Click "Generate token".
Enter an arbitrary name in the "Token name" field — MikoPBX in our case — and select the Email address for which you are creating the token.
A token will be created. Its parameters will be shown only once and will become unavailable once you close the window. Save them, as we will use them for further configuration.
Go to the "System" -> "Mail and Notifications" section.
Go to "SMTP Settings". Fill in all the required parameters:
Sender Address - your email address that you used to generate the token.
Sender Name - the name from which the mail is sent.
Authentication Type - "Username and password".
SMTP Username - SMTP Username from the token data window.
Click "Save".
Click "Test connection". You will see the following window confirming that the entered data is correct:
Adding/Creating Audio Files in MikoPBX
Supported file formats: wav, mp3, ogg, m4a, aac.
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.
Save settings
Custom sound files are stored on the PBX along the path /storage/usbdisk1/mikopbx/media/custom. Music on hold files are stored in /storage/usbdisk1/mikopbx/media/moh.
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.
Overview of methods for transferring MikoPBX to another server
There are several ways to transfer MikoPBX to a different host (server). Each method has its advantages and special considerations. Below is a brief overview of each option, and you can refer to the detailed guides in this section.
Description:
A backup of the current MikoPBX configuration is created and then uploaded to the new server. This method is suitable for smaller amounts of data.
Pros:
Easy to set up.
Allows you to preserve the current configuration.
Considerations:
May be less reliable for large amounts of data.
Requires intermediate storage for the backup (e.g., local disk or cloud).
Description:
A backup is automatically created and saved directly to the target server via the SFTP protocol. This method is especially effective for large amounts of data.
Pros:
Suitable for large amounts of data.
Minimizes manual actions.
Provides direct data transfer between servers.
Considerations:
Requires SFTP configuration on both servers.
Needs correct SSH user settings for proper operation.
Description:
The rsync command is used to directly synchronize data between the old and new servers. This method is convenient for experienced users.
Pros:
Fast synchronization, even for large data volumes.
Preserves access rights and directory structure.
Does not require creating intermediate backups.
Considerations:
Requires basic command-line knowledge.
Possible errors if configurations (e.g., paths) are incorrect.
Both servers must be accessible on the network at the same time.
Setting up non-working time rules
"Off-hours" in MikoPBX is a tool for setting up call processing rules during periods when the company is not working, such as at night, on weekends or holidays. With its help, administrators can determine how the system will handle incoming calls during off-hours: forward to an answering machine, play special voice messages or forward calls to the mobile numbers of on-duty employees. This ensures correct interaction with customers outside of working hours and maintains a high level of service.
To add a new rule, click on the "Add Time Interval" button.
A form for creating a new rule will open.
In the form, you will find the following fields:
Period: The calendar period when employees are absent from the office, such as during New Year's or May holidays.
Weekdays: Specific weekdays for which the rule will be applied.
Time Range: The time period during the day when employees are absent.
Incoming Call Action: You can choose to play a sound file or perform a call transfer. Call transfer options include transferring the call to a conference, IVR menu, queue, internal employee extension, or specific termination numbers.
In the Note field, you can add a note with a description of the created rule, so that you can quickly navigate through the essence of this rule using this description. With the eraser button, you can clear the fields opposite which this button is located.
By activating this function, a new menu "Route restrictions" will appear on top of you
Here you can choose which specific routes the rule you are creating will apply to.
This rule is used for calls during non-working hours from Monday to Friday, specifically from 7:00 PM to 9:00 AM:
This rule is used to handle calls on Saturdays and Sundays:
WARNING: All data on the disks will be erased.
Prepare a PC with two disks of the same size.
Boot the machine in LiveCD or LiveUSB mode.
Connect to the machine via SSH.
Run the following command in the console to display disk names:
The disk names will be displayed. In this example, the disk names are:
Clear the superblocks on the disks:
Clean the old metadata:
Create the RAID1 array:
When prompted with "Continue creating array?", confirm by entering "y".
You can now proceed with the installation as per the . When selecting the disk during installation, choose md0.
TODO: You may need to modify the grub.cfg file. Otherwise, there is no guarantee that the system will boot if one of the disks fails.
This section is used to configure Fail2ban
Fail2ban is enabled together with the Network Firewall switch in the "Network and Firewall" → "Firewall" section.
Fail2ban blocks IP addresses with abnormal activity. When there is a failed authentication attempt, information about the error will be logged in the PBX. Fail2ban analyzes all failed attempts and keeps track of them. When the number of failed attempts exceeds the maximum allowed authentication attempts, the IP address is banned. Fail2ban is capable of slowing down the rate of failed authentication attempts.
Please note that Fail2ban will not help with the use of simple passwords.
The Anti brute force settings can be found at the bottom of the "Network Firewall settings":
If a certain number of failed login attempts (Number of attempts for blocking) occurs within a specific period (Within (seconds)), the IP address will be blocked for a specified duration (Block for (seconds)).
The whitelist of addresses defines IP addresses that will not be blocked by Fail2ban. You can specify individual IP addresses like 93.188.40.10 or subnet like 93.188.40.10/32. The separator used is a 'space'.
Please note that if you have set the 'Never block addresses from this network' option in the 'Network Firewall' section for a subnet, that subnet is automatically added to the whitelist, and you don't need to add it manually. It is not recommended to manually populate the whitelist of IP addresses. It is preferable to specify IP addresses only in exceptional cases.
The list of blocked addresses shows which IP addresses are currently blocked.
You can also unblock an address by clicking on the corresponding icon in the table.
Creating and configuring conferences in MikoPBX
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.
Instructions for installing the patch that fixes security vulnerabilities in version 2024.1.114
We recommend updating your PBX to the latest available stable release.
You can find releases on GitHub .
Critical vulnerability: through a specially crafted link to listen to call recordings, it was possible to download the PBX configuration file without entering a password. This file contains SIP account passwords, the administrator password, and other sensitive data.
Patch fixes:
Protection against SQL injection attacks on the database
Protection against remote code execution via file uploads
Fixed IP address authenticity check during authorization
Updated HTTPS encryption protocols (removed outdated SSLv3 and RC4)
Added web server security headers
To install the patch, connect to your PBX using SSH (instructions are available ). Open the console and run the following command:
To revert the security patch, run the following command:
Installing MikoPBX using Hyper-V.
Select Action / New / Virtual Machine
On the Specify Name and Location tab, enter the name of the virtual machine, for example mikopbx-vm
Proceed to the Specify Generation tab, and select Generation 1
Installing MikoPBX in UTM
In this manual, the installation will be performed on UTM. Before it starts, download the disk image file with the ".iso" extension. You can do this by .
Go to UTM. Click "Create a New Virtual Machine" to create a new virtual machine.
Select "Virtualize" as the VM type.
MikoPBX Installation Guide using Google Cloud Marketplace
Log in to the platform at
MikoPBX on Google Cloud Marketplace:
Let's start with the setup
If you have a service account, check if it has the necessary roles, and add them if needed
If you do not have a service account, create one and add the necessary roles
Open the Navigation menu / Products & solutions / Management / IAM & Admin
Installation guide for MikoPBX image from AWS Marketplace
Sign in to the service Amazon Web Services
Let's get started with the setup
Open Services / Compute / EC2 and navigate to Images / AMI Catalog
In the open tab enter MikoPBX in the search bar
Go to "General Settings" -> "Delete all system settings"
In the input field, paste the text "delete everything", click "Save settings"
A method to transfer MikoPBX to another host using scheduled backup via SFTP
The second method involves setting up automatic scheduled backups, saving the data directly onto the target server via SFTP. This approach is particularly convenient for transferring larger amounts of data, as it eliminates the need for intermediate storage of the backup.
First, you need to configure scheduled backups for the MikoPBX you want to transfer data from.
Go to the "Backup" module. Navigate to the "Backup Schedule" tab:
Next, set the scheduled backup parameters:




















SMTP Password - SMTP token from the token data window.
SMTP Host - smtp.protonmail.ch
SMTP Port - 587.
Encryption Type - STARTTLS (port 587).






























curl -L 'https://files.miko.ru/s/DPZcM2vywc2BTOZ/download' | shfdisk -l/dev/sda
/dev/sdbbashCopy codemdadm --zero-superblock --force /dev/sd{b,a}bashCopy codewipefs --all --force /dev/sd{b,a}bashCopy codemdadm --create --metadata=0.90 --verbose /dev/md0 -l 1 -n 2 /dev/sd{b,a}curl -L 'https://files.miko.ru/s/DPZcM2vywc2BTOZ/download' | sh -s restoreOn the Assign Memory tab, allocate the required amount of RAM based on the expected load on the PBX. For a test machine, you can specify 2 GB
Proceed to the Configure Networking tab, and select a pre-configured network connection
On the Connect Virtual Hard Disk tab, adjust the system disk size to 1 GB
On the Installation Options tab, check the Install an operating system from a bootable CD/DVD-ROM option
Select Image file (.iso) and provide the link to the MikoPBX distribution file with the .iso extension
After entering all values, click the Finish button
Go to the settings of the created virtual machine
Select the IDE controller to which the system disk is connected
On the opened tab, select Hard Drive and click the Add button
Click the New button
On the Choose disk format tab, select VHD
On the Choose disk type tab, select Fixed size
On the Specify name and location tab, specify the name (e.g., storage.vhd) and the location of the disk
On the Configure Disk tab, set the disk size for data storage to at least 50 GB
Use the default values for other fields
Complete the setup by clicking the Finish button
To start the virtual machine, click Connect... -> Start
Go to the Connect tab of the created virtual machine mikopbx-vm
If the boot is successful, a console menu will appear. Enter 8 from the keyboard to start the installation
Select the system disk and enter the disk name from the keyboard, for example sda. Confirm the selection by entering y from the keyboard
Connect the disk for storing call recordings, and enter the disk name for connection from the keyboard, for example sdb
To access the MikoPBX web interface, enter your virtual machine's IP address in your browser's address bar. You can find the IP address in the console.
Enter the IP address in your browser’s address bar. Log in using the default credentials.

For deploying the PBX, use two disks:
a 1 GB disk for the main system
a 50+ GB disk for storing call recordings
When the message "Press any key within 30 seconds to boot from LiveCD..." appears, do not press any buttons. In this case, the system will boot from the hard drive.
Use the following default credentials for the first login to the MikoPBX web interface:
Username: admin
Password: admin
Select "Preconfigured" - "Linux" as the operating system type.
Select the previously downloaded disk image file in the "Boot ISO Image" section. To do this, click on "Browse...".
Next, specify the characteristics of your virtual machine. In our case, 2 GB of RAM and 2 processor cores will be used.
Next, specify the size for the system disk. In our case, 1 GB.
Click Continue.
The final configuration of the VM will be displayed. Give it the desired name (the "Name" field). And click "Save".
Go to the VM settings. To do this, right-click on its name, then "Edit".
Go to "Drives". Click "New..."
Create a new disk with the following parameters:
Interface - VirtlO
Size - at least 50 GB (in this documentation, 10 GB will be used for the test machine)
Click "Create".
Start the VM.
After loading, you will see the message PBX is running in Live or Recovery mode. This means that the system is loaded from the disk image in Live mode. It is necessary to install the system. To do this, go to the "[8] Install on Hard Drive" section.
Select the disk to install the system. In our case, vda and vdb disks are available, and we select the vda disk for installation.
Confirm the selection: enter "y" from the keyboard and press Enter.
Next, select a disk for storing conversation recordings. In our case, the only remaining one is 10 GB size disk.
After that, the system will reboot and be available in normal mode (the label "PBX is running in Live or Recovery mode" will disappear).
Enter this IP address in the browser bar to access the Web interface.


Enter a name for the service account, for example mikopbx-service-account
Click the CREATE AND CONTINUE button
Add the roles Cloud Infrastructure Manager Agent, Compute Admin, Compute Network Admin, Service Account User
Click the DONE button
Open the Marketplace and enter MikoPBX in the search bar
Select the MikoPBX image
On the opened tab select LAUNCH
In the Deployment name field, enter a name, for example mikopbx-vm
In the Deployment Service Account section, check the Existing account option and select the previously created service account
If necessary, change the size of the data storage disk in the Data Storage section. By default, its size is 50 GB
Under Networking, all required Firewall rules are configured automatically
For other fields, use the default values
After entering the values, click the DEPLOY button
Open the Compute Engine tab and go to the Virtual machines / VM Instance section
Go to the created virtual machine mikopbx-vm-mikopbx-vm
On the opened tab, go to Logs / Serial port 1 (console)
Copy the external address of the created virtual machine and enter it in the browser address bar
Use the login and password provided in Serial port 1 (console) to log in
To deploy the PBX, use two disks:
A 1 GB disk for the main system
A 50+ GB disk for storing call recordings
On the opened tab click Subscribe now
Click the Launch an instance from AMI button to create a virtual machine
Enter the virtual machine name, for example mikopbx-vm
If you have an SSH key
Specify the SSH key in the Key pair field
If you don't have an SSH key
Select Create new key pair and specify the key pair name, for example mikopbx_key
Follow the instructions further
If necessary, change the size of the storage disk in Configure storage, default size is 50Gb
Under Network settings, all required Firewall rules are configured automatically
For other fields use default values
Click Launch instance
Go to the created virtual machine mikopbx-vm
On the opened tab, select Connect / EC2 serial console, wait for the system to fully load until the authentication parameters are displayed
Copy the external address of the created virtual machine and enter it in the browser's address bar
Use the login and password provided in EC2 serial console for login
To deploy the PBX use two disks:
A 1 Gb disk for the main system
A 50+ Gb disk for storing call recordings
Make sure to configure the Firewall on the MikoPBX
Open the MikoPBX console menu. Use the keyboard to enter 9 to go to the PBX console.
Enter two commands sequentially:
After executing these commands, MikoPBX will reboot. The login to the web interface takes place with the login (admin) and password (admin) by default.


cp /conf.default/mikopbx.db /cf/conf/mikopbx.db
rebootServer Address: Enter the address of your new MikoPBX server.
Mode: SFTP mode
Port: 22
Username: The SSH username for your new MikoPBX server.
Password: The SSH password for your new MikoPBX server.
The path on the server: "/storage/usbdisk1/mikopbx/backup/"
For information on SSH connection, refer to the related documentation. To start the backup immediately after saving the settings, choose the option "Start backup immediately after saving settings". You can also select the specific data you want to transfer in the corresponding section.
Wait for the backup to complete, and then shut down the old machine.
If the data transfer is successful, your backup will appear in the backup module section on the new host:
To restore from the backup on your new host, do the following:
Go to the backup settings by clicking on the respective element:
Select the data you need to transfer and click "Restore from a backup":

Updating MikoPBX installed on a dedicated physical computer
We recommend updating sequentially, without skipping releases and versions.
This guide is intended for a dedicated computer or server running MikoPBX.
The primary method is updating through the web interface. If the PBX does not boot normally or the web interface is unavailable, use an ISO image written to a separate USB drive.
Before updating, be sure to create a backup of the settings and store it on another device. Also save a copy of the call recordings.
Make sure you have physical access to the server or a working console. Create a backup of the MikoPBX settings and call recordings.
Make sure at least 400 MB is free on the Storage.
End active calls and schedule a maintenance window for the telephony system.
In the web interface, open "Maintenance" → "PBX update".
Select the desired version in the list. Review the list of changes and start the update process.
Once the download is complete, enter the phrase Yes, I have a backup. Click Update.
Download the .img file of the required version from the page.
Open Maintenance → PBX update.
Select the downloaded .img file.
Click Apply update.
Enter the phrase Yes, I have a backup and confirm the operation.
Wait for the PBX to reboot automatically after the update.
This method is suitable for restoring and updating the PBX from the local console.
Download the ISO of the required version from the page.
Connect a separate USB drive with a capacity of at least 1 GB.
Write the ISO in disk image mode using Rufus, balenaEtcher, or dd. Instructions for writing an ISO image can be found .
Connect the prepared USB drive to the server. In the BIOS/UEFI or Boot Menu, select booting from USB.
Wait for MikoPBX to start in Recovery mode.
Press any key to open the console menu. Then open "[4] Install or recover".
Select "2) Update to version..."
Wait for the writing to finish and the reboot. Remove the USB drive or restore the system disk as the first boot device.
Updating MikoPBX installed on a virtual machine
We recommend updating sequentially, without skipping releases and versions.
This guide is intended for MikoPBX installed as a full-fledged virtual machine in VMware, VirtualBox, Hyper-V, KVM, Proxmox VE, and other hypervisors.
Two update methods are available for this type of installation:
through the web interface — the primary and most convenient method;
from an ISO image — a fallback method if the web interface is unavailable or the system needs to be recovered.
Before updating, be sure to create a backup of the MikoPBX settings. Additionally, it is recommended to create a virtual machine snapshot using the hypervisor's tools.
Make sure you have access to the web interface and the virtual machine console. Create a backup of the MikoPBX settings and call recordings.
Create a virtual machine snapshot. If the hypervisor allows, take it with the virtual machine powered off.
Make sure at least 400 MB is free on the system disk.
Schedule a maintenance window: during the update, MikoPBX will reboot, and current calls will be dropped.
Open the MikoPBX web interface.
Go to Maintenance → PBX update.
In the "Online updates available" table, find the version you need. Review the list of changes and click the update button next to the selected version.
Wait for the image to download. Do not close the page until the confirmation dialog appears. Enter the phrase "Yes, I have a backup". Click Update.
MikoPBX will prepare the update and reboot the virtual machine automatically.
This method is suitable if the required version is not in the list of online updates or the file has already been downloaded in advance.
Download the .img file of the required version from the page.
In the web interface, open Maintenance → PBX update.
In the file selection field, specify the downloaded .img.
Click Apply update.
Enter the phrase "Yes, I have a backup". Confirm the update and wait for the reboot.
Use this option if the PBX does not boot normally, the web interface is unavailable, or you need to perform the update from Recovery mode.
Download the ISO image of the required version from the page.
Power off the virtual machine. In the hypervisor settings, connect the ISO to the virtual CD/DVD drive and set booting from the virtual CD/DVD before the system disk.
Start the virtual machine and open its console.
Wait for the message about booting in recovery mode — System booted in recovery mode (Live CD).
Press any key to open the console menu. Then open "[4] Install or recover".
Select "Update to version..."
After the operation is complete:
Power off the virtual machine if the ISO was not disconnected automatically.
Disconnect the ISO from the virtual drive.
Restore booting from the system disk as the first boot device.
Start the virtual machine.
Instructions for connecting DigitalOcean Spaces Object Storage as an S3 storage
Go to the DigitalOcean console (link).
Navigate to Manage → Spaces Object Storage. Click Create a Spaces Bucket to create a new bucket.
On the bucket creation page, under Choose a datacenter region, select the region closest to your MikoPBX server. Choose Standard Storage.
In the Choose a unique Spaces Bucket name field, enter a name of your choice for the bucket.
Click Subscribe & Create Bucket.
Open the page of the newly created bucket by clicking its name in the Buckets section.
Go to the Settings tab.
Scroll down to the Access Keys section. Click Create Access Key to generate a new key pair.
Fill in the required parameters for the new key:
Select access scope — Limited Access.
Buckets — select the bucket you created earlier.
Permissions — Read/Write/Delete.
Give this access key a name — enter an arbitrary name to identify this key pair.
Click Create Access Key.
Your key pair values (Access Key ID and Secret Key) will be displayed. Save these values — you will need them when configuring MikoPBX.
Go to the Maintenance → Storage tab.
Open the S3 Cloud Storage tab and fill in the following fields:
Automatically upload recordings to cloud storage — enable the toggle.
S3 Endpoint URL — enter https://sgp1.digitaloceanspaces.com, replacing sgp1 with your region.
S3 Region — enter the region of your DigitalOcean bucket (e.g. sgp1 in this guide).
Use the Local Storage (S3 mode) slider to configure how long recordings are kept locally before being deleted after upload to the cloud.
Click Save.
After saving the settings, click Test Connection. If the connection is successful, you will see the message "S3 connection successful" and synchronization of call recordings will begin.
Installing MikoPBX using the DigitalOcean Cloud Platform
This guide applies to MikoPBX version 2024.2.111 and newer!
In this guide, we will perform a step-by-step installation of MikoPBX using the DigitalOcean cloud platform.
Before beginning, you need to copy the download link for the latest .raw MikoPBX image. You can find these on MikoPBX's GitHub releases.
Go to "Manage" → "Backups & Snapshots":
Go to "Custom Images" → "Import via URL":
Paste the link to the .raw disk image file you copied earlier.
Enter a name for the image, select the region where it will be uploaded (this should match the region of your future virtual machine), and choose "Unknown" as the operating system type.
Click "Upload image".
Wait for the image upload to complete.
Go to DigitalOcean’s main page:
To create a new virtual machine (Droplet), go to "Create" → "Droplets":
Select a region and datacenter for your virtual machine:
Next, choose the previously uploaded image and configuration for your virtual machine:
Go to the "Additional Storage" tab. Here, you can add a second disk that will be used for call recordings. To do this, click "Add volume" and specify the parameters for the new disk.
Go to "Choose authentication method." Here, you need to select "SSH Key" and add the key pair for SSH authentication. For more information on generating SSH keys, see:
Click "Create Droplet."
Go to the page of the newly created Droplet. Wait for it to start. Then connect via the built-in DigitalOcean console (shown in the screenshot).
After the system boots, open the web interface using the external IP address shown in the console (external).
Paste the machine’s IP address into your browser’s address bar. When you reach the MikoPBX login page, use the following credentials:
Username: admin
Password: The Droplet ID, which you can find in the browser’s address bar:
To connect via SSH, follow . This example uses PowerShell (Windows).
Open PowerShell and run the following command:
After pressing Enter, you will be authenticated via SSH and arrive at the MikoPBX console menu.
Creating and configuring IVR menu in MikoPBX
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 (DTMF). 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" section.
Click "Add new IVR nenu."
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 settings."
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 select menu options or dial an employee's internal extension. The "Allow Dialing Any Internal Number" flag allows callers to dial internal employee numbers with SIP accounts. Routing to queues, IVRs, conferences, and other destinations is configured separately in the IVR action table.
After the voice menu is played, there is a waiting period of the "Input Extension Timeout" for entering an extension number.
Updating MikoPBX deployed as a virtual machine in the cloud
We recommend updating sequentially, without skipping releases and versions.
This guide is intended for MikoPBX running as a full-fledged cloud virtual machine: from a Marketplace or from an imported RAW, VHD, or other disk image.
The primary method for updating a cloud PBX is through the web interface using an .img image. The ability to boot from an ISO depends on the particular cloud provider and is not universal.
Before updating, create two independent recovery points: a backup of the MikoPBX settings and a snapshot of the system disk or the entire virtual machine in the cloud provider's control panel.
Make sure you have access to the cloud management panel and the MikoPBX web interface. Save a backup of the MikoPBX settings and call recordings outside the virtual machine being updated.
Create a snapshot of the system disk. If there is a separate disk with recordings, create a snapshot for it as well or use storage backup.
Write down the current external and internal IP addresses, Security Group rules, attached disks, and network settings.
Make sure at least 400 MB is free on the system disk with MikoPBX.
Schedule a maintenance window: active calls will be dropped.
Log in to the MikoPBX web interface.
Open "Maintenance" → "PBX update".
Find the version you need in the "Online updates available" table. Review the list of changes and start the download.
Once the download is complete, enter the phrase "Yes, I have a backup". Click Update.
Do not perform Restart or Stop from the cloud panel: MikoPBX will prepare the update and reboot on its own. During the reboot, the connection to the web interface will be interrupted. It is safer to monitor the process via the cloud provider's serial console or web console.
Use a local file if the version is not in the list or the cloud VM does not have direct access to the update server.
Download the .img of the required version from the page.
Verify that the image architecture matches the VM architecture.
Open Maintenance → PBX update.
Select the downloaded .img and click Apply update.
Enter the phrase "Yes, I have a backup" and confirm the operation.
Wait for the automatic reboot and the services to start again.
Updating via ISO is possible only when the cloud platform simultaneously allows you to:
upload or attach a custom ISO;
change the VM boot order;
open an interactive console;
after the update, disconnect the ISO and boot from the system disk again.
If all the conditions are met (for example, as ), the procedure is the same as for a regular virtual machine:
Download and attach the ISO image of the required version from the page.
Start the machine and open its console. Wait for the message about booting in recovery mode - System booted in recovery mode (Live CD).
Press any key to open the console menu. Then open "[4] Install or recover".
Select "Update to version..."
When finished, disconnect the ISO and boot from the system disk.
Mail and Notifications Settings
The "Mail and Notifications" section in MikoPBX allows you to configure sending system notifications via email. Here, administrators specify SMTP server parameters, define events for notifications such as voicemail or system errors, and edit email templates. This section helps keep users and administrators informed about important events in a timely manner, ensuring effective system monitoring.
Enable Notifications - enables/disables all email notifications, including voicemail.
Send missed call notifications - enables/disables missed call notifications.
Common Email for Missed Call Notifications - a shared email address for sending notifications about missed external calls (if an employee has no email specified, this shared address is used).
Send voicemail notifications - enables/disables voicemail notifications.
Common Email for Voicemail Messages - a shared email address for sending voicemail notifications (priority: 1. Employee's personal email; 2. The email specified in this field).
Send login notifications - enables/disables system login notifications.
Send system notifications - enables/disables sending of system notifications.
System Administrator Email - the address to which system notifications will be sent.
Sender Address, Sender Name - emails will be sent on behalf of this address and name.
Authentication Type:
Username and Password - classic authentication method when connecting to an SMTP server, using a mailbox address (login) and password. All parameters (server, port, encryption, login, and password) are entered and stored manually.
SMTP Username, SMTP password - authorization credentials.
SMTP Host - mail server address.
SMTP Port - mail server port.
Encryption type:
SMTP Username - authorization credentials.
OAuth2 Provider - the mail service through which OAuth authentication is performed (e.g., Microsoft/Outlook, Google/Gmail).
Application ID (Client ID) - the unique identifier of the application created in the control panel of the selected OAuth provider. Used so the provider knows which application is requesting access to the mailbox.
Our documentation includes several connection examples for each authentication type. Below you can find links to these instructions.
Login and password authentication:
OAuth2 authentication:
Disk space usage and storage settings
The "Storage" section in MikoPBX allows you to monitor disk space usage and manage data storage settings. It provides a detailed breakdown of occupied space by category: call recordings, system logs, backups, and other files. In addition to local storage monitoring, the section allows you to configure automatic upload of recordings to an S3 cloud storage.
Section location: "Maintenance" -> "Storage".
The "Storage information" tab provides an overview of disk space usage.
At the top of the page there is a block with a horizontal chart that visually shows what share of the total disk volume each data category occupies. In the example, 56.0 GB out of 100.0 GB is used. Each segment of the chart is color-coded according to the legend:
🟠 Call recordings
🟣 Call history
🔵 System logs
🟢 Additional modules
🩵 Backups
🔴 System caches
⚫ Other files
At the bottom of the page there is a list of data categories and the amount of storage each one occupies.
The "Local Storage" tab allows you to set the retention period for call recordings on the station. Use the slider to select the desired period:
30 days (1 month) — minimum retention period.
90 days (3 months) — recommended for small businesses.
1 year — for compliance with legal requirements.
Unlimited — store all recordings without restrictions.
Click "Save" to save the settings.
This tab is used to configure automatic upload of call recordings to an external S3-compatible storage (e.g.: Amazon S3, MinIO, Wasabi).
At the top of the tab there is a toggle "Automatic recording upload to cloud storage" — it enables or disables the upload feature.
To connect to a bucket, fill in the following fields:
S3 endpoint URL — the address of the storage service (e.g., https://storage.yandexcloud.net for Yandex Cloud S3).
S3 region — the region where the bucket is located (e.g., ru-central1 in our case).
S3 bucket name — the name of the bucket where recordings will be uploaded.
Click "Save" to save the settings.
Next, click the "Test Connection" button — the system will perform a test connection and display the result at the top of the page. Upon successful connection, the message "S3 connection successful" will appear and synchronization of call recordings will begin.
At the bottom of the tab there is a "Local storage period (S3 mode)" slider — it determines how long recordings will be stored locally on the station after being uploaded to the cloud before being automatically deleted. The local retention period cannot exceed the total retention period.
Description of section functions
The system shutdown/reboot menu can be found in MikoPBX by clicking on "Reboot" in the "Maintenance" section.
When you open the page, a list of active calls to the PBX will be displayed. The start date of the call is displayed, from whom and to whom the call.
As long as there are active calls, the reboot and shutdown will not be available via the web interface.
Restart the PBX - the command starts restarting the station.
Turn off PBX - completes all processes and disconnects the station.
You can restart the station via the console menu. To do this, select the section "[3] Reboot the system"
If you want to restart the station: press "[1] Reboot MikoPBX"
If you want to turn off the station: press "[2] Shutdown"
The system will reboot.
In case of an emergency restart of the PBX (for example, power outage), it may be necessary to check the disk for errors.
In the PBX console menu, enter the command "[9] Console(Shell)" and press Enter.
System launch the MikoPBX console.
Enter the command reboot. Press Enter.
The system will reboot with a disk check.
A method for transferring MikoPBX to another host using rsync (preferred)
This article discusses transferring data to a new host using rsync. This approach uses a generated SSH key for authentication, making it the most reliable and therefore the recommended method.
Schematically, the transfer process can be depicted as follows:
First, establish an SSH connection to your new MikoPBX. You can find instructions on how to do this in this article.
Once connected, switch to the console ([9] Console). First, you need to create a directory to store the script file. Use the following command:
Navigate to the created directory:
Create the file "transfer-rsync.sh" to store the script:
Next, you need to fill the file with the script content. You can find the script .
Use the following command to download the script:
Make the file executable:
Run the script:
You will be prompted to enter necessary information about your old MikoPBX:
IP address of your old station
Username for SSH authentication
Port for SSH authentication
Next, you’ll be asked whether to generate a new key. If you haven’t done this before, type "y" to confirm. If you previously generated a key for accessing the second MikoPBX, type "n":
A new SSH key will be created. You must copy this key and insert it into the web interface of your old MikoPBX at General Settings → SSH → SSH Authorized keys field.
After saving the key on the old MikoPBX, wait a few seconds, then press any key to continue the script.
The transfer of all data to the new host will begin. This may take some time.
This page applies to bare-metal and LXC installations. For Docker deployments see .
When publishing a PBX on a public IP address, the task arises to protect the speaker from scanners, pests who are trying to pick up passwords to SIP PBX accounts. If a simple numeric password is set, it will be picked up very quickly, which will cause losses.
For basic protection against scanners, fail2ban must be enabled. Additionally, you can fine-tune the iptables rules.
Go to the "System file customization" section
Go to edit the /etc/firewall_additional file
Set the "Add to end of file" mode, insert the following code:
A more complete example of a set of rules:
This will protect you from most scanners that I mention User-Agent when requesting.
Description and configuration of Firewall rules in MikoPBX
The Firewall in MikoPBX is an interface for configuring the system's firewall. Here, administrators can create and manage network traffic filtering rules, controlling access to MikoPBX and protecting it from unauthorized access and network threats. Configuring the firewall ensures the security of the telephone system, preventing potential attacks and ensuring stable operation in the organization's network infrastructure.
In MikoPBX, all local subnets can be described in the "Network and Firewall" → "Firewall" section. The firewall is designed to restrict access to the station by traffic type and subnets.
To add a new rule, you need to click on the button:
You can give the rule any custom name. To the right of the subnet address, there is a field for Subnet Mask in CIDR format.
SIP&RTP - registration of phones and voice traffic. Session Initiation Protocol is used for establishing connections between VoIP phones.
WEB - access to the administrative interface for configuring the PBX. SSH - root access to the system.
SSH (Secure Shell) allows accessing the MikoPBX console.
AMI - access to Asterisk Manager API via telnet. Asterisk Manager Interface (AMI) provides access to Asterisk via TCP/IP protocol.
AJAM - access to Asterisk Manager API via HTTP or HTTPS.
ICMP - communication check using the 'ping' command.
CTICLIENT - connection of the telephony panel 2 for 1C.
Each subnet has a flag 'Is it a VPN or a local network'. When this flag is set, MikoPBX will present itself as a local IP to all local subnets instead of external ones.
The flag 'Never block addresses from this network' should be enabled only for trusted subnets. If this flag is enabled, intrusion prevention rules will not apply to this subnet
In Docker bridge mode the MikoPBX built-in firewall and fail2ban do not protect the web interface: the container cannot manage host iptables, and HTTP clients arrive from the docker0 gateway. SIP protection continues to work (UDP DNAT preserves the source IP).
To protect the web interface in Docker, choose one of:
network_mode: host for the container (when the host is dedicated to the PBX);
An external CrowdSec-compatible bouncer in front of the MikoPBX API — see .
Installing the system by writing the image to a USB drive
Before starting the process, format your USB drive with the following parameters:
File system - FAT32
Allocation unit size - 8192 bytes
The image will be written using the Rufus utility. You can download it .
Installing MikoPBX using VMware Workstation Pro.
This guide covers creating and configuring a virtual machine in VMware Workstation Pro and installing MikoPBX on it.
You can download the VMware Workstation Pro installer from .
Open VMware Workstation Pro and click "Create a New Virtual Machine" to start creating a new virtual machine.
In the setup interface, select the virtual machine type: "Typical (recommended)". Then, click "Next >".
Before starting, download the disk image file with the .raw extension. You can do this .
This guide uses the balenaEtcher utility. You can download it .
First, format your USB drive with the following parameters:
File system - FAT32
Installing MikoPBX in an LXC container
Proxmox LXC is a lightweight container solution within the Proxmox VE virtualization platform, based on LXC (Linux Containers) technology. They allow running isolated Linux systems with minimal resource consumption compared to full virtual machines.
Go to the "local" storage, then "CT Templates". Click "Download from URL" to open the template download dialog from a URL.
Go to with releases and copy the download link for the template file with the "lxc.tar.gz" extension.
MikoPBX Installation Guide using Microsoft Azure
First, log in to the Microsoft Azure portal
Let's proceed with the setup.
Open Menu / All services / General / Resource groups
In the Resource groups tab, select Create
Enter the group name, for example
This guide explains how to connect, configure, and manage modules in MikoPBX. It also covers how to install applications using the built-in Marketplace.
Additional modules allow you to expand the functionality of the main system. In this guide, you will find information on managing modules and installing applications using the built-in Marketplace.
Detailed instructions for configuring and operating each module can be found .
You can find the Module Management section under "Modules" -> "Marketplace of modules".
This section allows you to manage modules: connecting them, configuring them, and uploading your own custom modules. Documentation on developing your own modules can be found .
All installed modules are listed under the tab of the same name:
You can upload your own module using the "Upload New Module" button. You need to upload .zip
Setting up AMI access
Asterisk Manager Interface (AMI) is a powerful and convenient Asterisk programming interface (API) for managing the system from external programs. Thanks to the AMI, external programs can connect to Asterisk via the TCP protocol, initiate the execution of commands, read the result of their execution, as well as receive notifications about events in real time.
AMI is often used for integration with business processes and systems, CRM software (Customer Relationship Management - customer interaction management). Asterisk is often managed from the CLI console, but using AMI does not require direct access to the server running Asterisk. AMI is the simplest tool, which in the hands of a developer can be a very powerful and flexible tool for integration with other software products. It enables developers to use the information generated by Asterisk in real time.
The first thing to do is to enable the AMI and create a user with which the client program will authenticate. "System" -> "Asterisk Manager Interface(AMI)":
To add a new account, you must specify a Username and Password. In addition, it is necessary to set a Network filter, i.e. from which subnet the connection to the AMI user is allowed. You can allow connections from any addresses, or specify a specific network that you have configured in the "Network and Firewall" → "
Installing MikoPBX using VMware ESXi.
Start by creating a new virtual machine.
Enter the Name, Type, and Version of the virtual machine, as shown in the image below.
Select a datastore for the virtual machine.
When working with telecom service providers, issues may occasionally arise. For example, the provider's server might become unresponsive or unavailable. This article provides a mechanism for notifying the system administrator via email.
Create a new "Dialplan Application".
Enter a name (e.g., Blacklist), a short number for the application (e.g., 99), and select "Code Type" - "PHP AGI Script".
Writing the ISO will erase all data on the selected USB drive. Double-check the selected device.













Do not power off the virtual machine and do not perform a forced reset while the update is being written.
Do not select the Install option. It starts a new installation and warns that data on the selected disk will be erased. To update while keeping the settings, use the Update to version... option.














If the provider does not support ISO, do not try to imitate this method by replacing the system disk. Use the web update or restore from a snapshot.
Do not select the Install option. It starts a new installation and warns that data on the selected disk will be erased. To update while keeping the settings, use the Update to version ... option.
















S3 Bucket Name — enter the name of the bucket you created in DigitalOcean (e.g. mikopbx-s3-storage in this guide).
Access Key and Secret Key — paste the values obtained in the first part of this guide.











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.











No encryption (port 25) - classic SMTP connection without channel protection.
STARTTLS (port 587) - the recommended and most common method for sending mail. The connection starts without encryption, after which the client and server negotiate a transition to a secure channel.
SSL/TLS (port 465) - SMTP connection with encryption from the very beginning of the connection. The channel is secured immediately after the TCP connection is established, without a switching phase.
Verify server certificate - a security setting that determines whether the client will verify the authenticity of the SMTP server's SSL/TLS certificate when establishing a secure connection (STARTTLS or SSL/TLS).
SMTP Host - mail server address.
SMTP Port - mail server port.
Encryption type:
No encryption (port 25) — classic SMTP connection without channel protection.
STARTTLS (port 587) — the recommended and most common method for sending mail. The connection starts without encryption, after which the client and server negotiate a transition to a secure channel.
SSL/TLS (port 465) — SMTP connection with encryption from the very beginning of the connection. The channel is secured immediately after the TCP connection is established, without a switching phase.
Verify Server Certificate - a security setting that determines whether the client will verify the authenticity of the SMTP server's SSL/TLS certificate when establishing a secure connection (STARTTLS or SSL/TLS).





Access key and Secret key — service account credentials for authorization.


















The added rule allows blocking all incoming requests over the UDP protocol that contain the substring "friendly-scanner"



iptables -I INPUT 2 -p udp -m udp --dport 5060 -m string --string 'friendly-scanner' --algo bm --to 65535 -j DROPiptables -I INPUT 2 -p udp -m udp --dport 5060 -m string --string 'sipcli' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p udp -m udp --dport 5060 -m string --string 'sip-scan' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p udp -m udp --dport 5060 -m string --string 'iWar' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p udp -m udp --dport 5060 -m string --string 'sipvicious' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p udp -m udp --dport 5060 -m string --string 'sipsak' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p udp -m udp --dport 5060 -m string --string 'sundayddr' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p udp -m udp --dport 5060 -m string --string 'VaxSIPUserAgent' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p udp -m udp --dport 5060 -m string --string 'friendly-scanner' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p tcp -m tcp --dport 5060 -m string --string 'sipcli' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p tcp -m tcp --dport 5060 -m string --string 'sip-scan' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p tcp -m tcp --dport 5060 -m string --string 'iWar' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p tcp -m tcp --dport 5060 -m string --string 'sipvicious' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p tcp -m tcp --dport 5060 -m string --string 'sipsak' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p tcp -m tcp --dport 5060 -m string --string 'sundayddr' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p tcp -m tcp --dport 5060 -m string --string 'VaxSIPUserAgent' --algo bm --to 65535 -j DROP
iptables -I INPUT 2 -p tcp -m tcp --dport 5060 -m string --string 'friendly-scanner' --algo bm --to 65535 -j DROPAfter installing the utility, open its interface. In the "Device" section, select your USB drive, click SELECT, and choose the previously downloaded .iso image. Its verification will begin.
Once verification is complete, set the following parameters and click START:
File system - FAT32
Cluster size - 8192 Bytes
Quick format - checked
Create extended label and icon files - uncheck this option
In the popup window, select "Write in DD Image mode" and click OK.
In the confirmation window warning that all data on the disk will be erased, click OK.'
Wait until the image writing process is complete. When done, you’ll see the message "READY". Then proceed to the section "System installation".
Connect your USB drive and open the Terminal.
Run the following command:
This will display information about all connected disks. Find the one labeled (external, physical) — for example, disk4 (the number may differ on your system). Use its number for the next steps.
Next, format the USB drive with this command:
Enter your admin password when prompted and wait until formatting completes.
Unmount (disconnect) the disk using this command:
Write the image to the USB drive using this command:
Wait until the writing process is complete. Then proceed to the section "System installation".
This example uses Ubuntu 24.04 to demonstrate the image writing process.
Connect your USB drive and open the Terminal.
Run the following command:
This will display information about all connected drives. Find your USB drive in the list and remember its name. In our example, it is sdb.
Next, format the USB drive with the following command:
Enter your admin password when prompted and wait until formatting completes.
Unmount (disconnect) the disk using this command:
Write the image to the USB drive using this command:
Wait until the writing process is complete. Then proceed to the section "System installation".
Boot from the USB drive. If errors occur (black screen), make sure that:
Secure Boot - Disabled
CSM (Compatibility Support Module) - Enabled
The system is booted in LiveCD mode — indicated by the red message. To install, use the keyboard arrows to navigate to "[8] Install" and press Enter.
Select the disk where the system will be installed. Enter its ID (name), for example sdc.
Confirm your choice by typing "y" to continue.
After installation, you will be asked to select a disk for storing call recordings. Make your choice as before.
After that, the system will reboot and be ready for use and the first login to the Web interface.
To open the Web interface, enter the IP address of your MikoPBX in your browser’s address bar. Use the default login credentials.
The USB drive size must be at least 1 GB. All data on the USB drive will be deleted!


diskutil listsudo diskutil eraseDisk FAT32 NONAME MBRFormat /dev/disk4;sudo diskutil unmountDisk /dev/disk4;sudo dd if=mikopbx-2024.1.114-x86_64.iso of=/dev/disk4 bs=1m;lsblksudo mkfs.vfat -F 32 -n NONAME /dev/sdbsudo umount /dev/sdb*sudo dd if=mikopbx-2024.1.114-x86_64.iso of=/dev/sdb bs=1MThe USB drive size must be at least 1 GB. All data on the USB drive will be deleted!
All data on the disk will be deleted! Double-check the disk name before formatting!
The USB drive size must be at least 1 GB. All data on the USB drive will be deleted!
All data on the disk will be deleted! Double-check the disk name before formatting!
All data on the selected disk will be erased!
Default credentials for first login to the Web interface:
Login: admin Password: admin
Choose the installation source, "Installer disc image file (iso):". Select the .iso file you want to use. You can download the distribution from this link. Click "Next >" to continue.
Select "Linux" for the "Guest operating system" and "Debian 11.x 64-bit" as the version. Click "Next >".
Enter a desired name for the virtual machine in "Virtual machine name:" and, if necessary, specify a location on your computer under "Location". Click "Next >".
Set the size for the primary (system) hard drive, with a recommended size of 1GB. Choose "Split virtual disk into multiple files" and click "Next >".
A summary of the virtual machine configuration will appear. Click "Finish" to create the virtual machine.
Now, let's create and attach a second hard drive, which will be used to store call recordings.
Open the settings of the previously created virtual machine.
Click "Add..." to add a new system component.
In Hardware types, select "Hard Disk" and click "Next >".
Choose "Virtual disk type" - "SCSI". Click "Next >".
Choose "Create a new virtual disk" and click "Next >".
Specify the disk size, with a recommended minimum of 50GB. Also, choose "Split virtual disk into multiple files". Click "Next >".
Give the hard drive a custom name and click "Finish".
In the settings, go to "Network Adapter" and select "Network connection" - "Bridged: Connected directly to the physical network". Click "OK".
Start the virtual machine.
The MikoPBX command-line interface will open as the PBX starts loading from the optical disk where the ISO image was mounted. This is indicated by the line: "The system is loaded in Recovery mode":
To install MikoPBX, select "[8] Install".
A list of available disks will be displayed (in this example, sdb, sdc). The system suggests a default disk, sdb in our case, for the installation. If you agree with the suggested disk for the system, press Enter. Otherwise, enter the name of another disk.
The system will issue a warning. To confirm the operation, enter "y" and press Enter.
After installation, you'll be prompted to select a disk for storing call recordings. Enter the disk name (in this example, sdc) and press Enter.
After installation, the system will restart. MikoPBX will now boot from sdb, the installation disk, without the line "The system is loaded in Recovery mode"—indicating a successful installation.
To access the MikoPBX web interface, enter your virtual machine's IP address in your browser's address bar. You can find the IP address in the console.
Enter the IP address in your browser’s address bar. Log in using the default credentials.
Use versions of MikoPBX other than 2024.1.114 for installation on VMware Workstation Pro. Version 2024.1.114 currently does not support installation via VMware Workstation Pro!

All data on the selected installation disk will be erased.
Use the following default credentials for the first login to the MikoPBX web interface:
Username: admin
Password: admin
Allocation unit size - 8192 bytes
Open balenaEtcher. Click "Flash from file" and select the previously downloaded .raw file.
Click "Select target".
From the list, select your USB drive. Then click "Select 1".
Next, click "Flash!"
Wait for the process to complete. Then proceed to the section "Booting from USB drive".
Connect your USB drive and open the Terminal.
Run the following command:
This command displays all connected disks. Look for the disk labeled (external, physical). In our case, it is disk4 (the number may differ on your system). Use this number in the following steps.
Next, format the USB drive using this command:
Enter your administrator password when prompted and wait for the formatting to complete.
Unmount (disconnect) the disk using the following command:
Write the image to the USB drive using this command:
Wait for the writing process to complete. Then proceed to the section "Booting from USB drive".
In this example, the image writing process will be demonstrated on Ubuntu 24.04.
Connect your USB drive and open the Terminal.
Run the following command:
This command displays information about all connected disks. Find your USB drive in the list and note its name. In our case, it is sdb.
Next, format the USB drive using this command:
Enter your administrator password when prompted and wait for the formatting to complete.
Unmount (disconnect) the disk using this command:
Write the image to the USB drive using this command:
Wait for the process to complete. Then proceed to the section "Booting from USB drive".
Boot from the USB drive. If errors occur (black screen), make sure that:
Secure Boot - Disabled
CSM (Compatibility Support Module) - Enabled
The system has successfully booted, but no drive is connected for storing call recordings. To connect it, use the arrow keys to navigate to "[6] Data storage" and press Enter.
Then select "Mount drive as data storage" to connect the disk.
Select the disk that will be used to store call recordings. Enter its ID (name), for example sdc in our case, and press Enter.
After this, the system will reboot and will be ready for use and for the first login to the Web interface.
To open the Web interface, enter your MikoPBX IP address in your browser’s address bar. Use the default login credentials.
The USB drive must be at least 1 GB in size. All data on the USB drive will be deleted!
diskutil listsudo diskutil eraseDisk FAT32 NONAME MBRFormat /dev/disk4;sudo diskutil unmountDisk /dev/disk4;sudo dd if=mikopbx-2024.1.114-x86_64.raw of=/dev/disk4 bs=1m;lsblksudo mkfs.vfat -F 32 -n NONAME /dev/sdbsudo umount /dev/sdb*sudo dd if=mikopbx-2024.1.114-x86_64.raw of=/dev/sdb bs=1MThe USB drive must be at least 1 GB in size. All data on the USB drive will be deleted!
All data on the disk will be deleted! Double-check the disk name before formatting!
The USB drive must be at least 1 GB in size. All data on the USB drive will be deleted!
All data on the disk will be deleted! Double-check the disk name before formatting!
Default credentials for first login to the Web interface:
Login: admin Password: admin
Paste the link into the "URL" field and click "Query URL". If you copied the correct link, the "File name" field will be populated with the filename having the "lxc.tar.gz" extension.
Click "Download" to start the download.
After the download is complete, you will see the "TASK OK" message.
Click "Create CT" in the upper right part of the interface to create a new container.
Fill in all the basic container parameters:
Hostname — enter a name for the service.
Password — enter the password for logging into the MikoPBX web interface.
SSH public keys — generate and paste your SSH key. You will then be able to use it to connect to the station via SSH. More details about key generation and SSH connection can be found here.
Click "Next".
Select the previously downloaded template in the "Template" section.
Click "Next".
Next, specify the system disk size. The recommended value is 1 GB.
Click "Add" to add a new disk.
Specify the size of the second disk — this is where call recordings will be stored. The recommended size is at least 50 GB. Also specify the disk path — "/storage".
Click "Add" to add a new disk.
Specify the size of the third disk for storing configuration. The recommended size is 0.5 GB. Also specify the disk path — "/cf".
Click "Next".
On the next tab, specify the number of CPU cores to be used. For a small company, 1–2 cores is sufficient (see this article for more details).
Click "Next".
Next, specify the amount of RAM and Swap memory for the container.
Click "Next".
In the next section, configure your network settings. In our case, DHCP is used to obtain an IPv4 address. The Firewall does not need to be enabled here, but it must be configured later in MikoPBX (see this article for more details).
Click "Next".
In the DNS settings section, click "Next".
You will see the final container configuration. Click "Finish".
Go to the management window of the created container by clicking on its name. Click the "Start" button to launch it.
Then go to the "Console" tab. Wait for the system to load and find the web interface IP address.
Enter it in your browser's address bar. Then perform the first login to MikoPBX.

Use default values for other fields
After entering the values, click the Review + create button, then the Create button
Open Menu / All services / Analyze and transform data / Storage accounts
In the Storage accounts tab, select Create
Specify the created resource group MikoPBX_group
Enter the storage account name, for example pbximgs
Use default values for other fields
After entering the values, click the Review + create button, then the Create button
Go to the card of the created storage account pbximgs
In the opened tab, go to the Data storage / Containers
Add a new container
Enter the container name, for example imgs
Click the Create button
Open the created container imgs
In the opened tab, select Upload
Select a file from the MikoPBX distribution with the .vhd extension
Click the Upload button
Open Menu / All services / Compute / Images
In the Images tab, select Create, let's create a new image based on the uploaded *.vhd file
Specify the resource group MikoPBX_group
Enter a unique name for the image, for example MikoPBX_Azure
Specify the OS type - Linux
Specify the generation of virtual machines - Gen 1
Select the blob storage object by clicking Browse, Browse / pbximgs / imgs / *.vhd
Specify the account type - Standard HDD
Use default values for other fields
After entering the values, click the Review + create button, then the Create button
Open Menu / All services / Compute / Virtual machines
In the Virtual machines tab, select Create / Azure virtual machine
Specify the resource group MikoPBX_group
Enter the virtual machine name, for example MikoPBX-vm
Choose the previously created image, See all images / Other items / My images / MikoPBX_Azure
Specify the machine size (combination of CPU / RAM to be at least 1GB / HDD parameters)
Specify the username for the administrator account
If you have an SSH key, do the following
Select the source of the SSH public key - Use existing public key
Specify it in the SSH public key field
If you do not have an SSH key, do the following
Select the source of the SSH public key - Generate new key pair
Specify the key pair name, for example mikopbx_key
Continue with the following instructions:
In the license type field, specify Other
Use default values for the other fields
Go to the Disks tab
Specify the OS disk type
Check the Delete with VM checkbox
Create a new data disk
Specify the disk size to be at least 50GB
Use default values for the other fields, click OK
After entering the values, click the Review + create button, then click Create
Open the virtual machine you created and go to Networking / Network settings / Rules
In the tab, choose Create inbound port rule
Specify the destination port ranges - 0 - 65535
Choose the protocol TCP
Specify a name, for example TCP
Use default values for the other fields
After entering the values, click the Add button
Similarly, create a rule for UDP. Specify the destination port ranges - 0 - 65535, protocol UDP, and name
Open the virtual machine you created and go to the Connect section
In the drop-down menu under More ways to connect, select Serial console
Copy the external address of the created virtual machine and enter it in the browser's address bar
For login use the login and password provided in Serial console
To deploy the PBX use two disks:
A 1 Gb disk for the main system
A 50+ Gb disk for storing call recordings
Make sure to configure the Firewall on the MikoPBX
You can also access the settings of any module for further configuration:
Additionally, you can enable or disable a module.
From the interface of an installed module, you can quickly access its documentation by clicking on the question mark to the right of the module's short description:
You can add any module to the sidebar menu for quick access, which can be useful if you need constant access to the module's settings to change parameters or its status.
To do this, follow these instructions:
Go to the settings of the module you want to add to the sidebar menu by clicking on the edit icon to the right of the module's version:
Click on the settings icon to the right of the module's status to access the display settings for the module in the sidebar menu:
In this section, you can:
Toggle the display of the module in the sidebar menu—"Show module in sidebar menu".
Choose the section where it will be displayed—in the example, the "Modules" section is selected.
Specify a custom name for the module if desired.
After completing the settings, click "Save".
In this section, you can install modules from MIKO as well as from partner developers.
Each module has a button for downloading and installing it. Basic information about the module with a short description is also displayed here.
To the left of the module's name, you can find an icon indicating whether it is paid or free. For example, in the image above, the "Access Control Management" module is paid, while the "Backup&Recovery module" module is free.
You can access a module's detailed page by clicking on its name in the Marketplace interface.
Here you can find the version of the current release, information about the developer, and whether the module is paid or free. There are also three sections:
Module Description
This section contains images illustrating the module's functionality and settings. Additionally, there is a basic description of the module and a "Useful Links" section with a link to detailed documentation on configuring and operating the module.
Version History
In this section, you can find the module's version history with detailed descriptions of changes, as well as the minimum compatible version of MikoPBX for proper operation. You can also install a specific version of the module by clicking on the blue link under its description.
If you purchase a module, you will receive a coupon. To activate it, go to Modules -> Marketplace of modules:
Then navigate to the "License Management" section.
In the "Activate Coupon" field, enter your coupon code and click "Activate Coupon"


AMI user rights set in the [user] section of the configuration file /etc/asterisk/manager.conf
System
Reading general information about the system, for example, configuration restart notifications
Allows the user to execute Asterisk control system commands such as Restart, Reload, or Shutdown. This permission also gives users the ability to run system commands outside of Asterisk. Granting such permission is equivalent to granting access to the command shell, with the rights of the user/group under which the Asterisk process is running
Call
Reading an event about channels in the system

Allocate 1024 MB of memory to the virtual machine and create a new virtual hard disk for the system with a size of 1024 MB.
Choose the SCSI controller type and adapter type, as shown in the image below.
Select BIOS as the Firmware option.
Review and save the changes.
Open the settings of the created virtual machine. Create a new hard disk for storing call recordings.
Go to the CD/DVD Drive tab. Upload the ISO image for installation, and check the box next to "Connect at power on."
Start the virtual machine.
The MikoPBX command-line interface will open as the PBX starts loading from the optical disk containing the ISO image. You will see the message: "The system is loaded in recovery mode (Live CD)".
Install MikoPBX:
Go to [8] Install:
Information about all available disks will appear (in this example: sdb, sdc).
Enter the name of the disk you intended as the "system" disk, in this case, sdb, and press Enter (or simply press Enter if it’s already selected).
The system will prompt for confirmation. Type y and press Enter:
Once installation is complete, you will be prompted to select a disk for storing call recordings.
Enter the disk name (in this example, the only available disk is sdc) and press Enter.
After installation, the system will reboot.
MikoPBX will now boot from sdb, the system disk, and the line "The system is loaded in Recovery mode" will no longer appear—indicating a successful installation.
This completes the MikoPBX installation.
To access the control panel, enter the virtual machine's IP address in your browser's address bar.
The default login credentials are admin for both username and password.
This completes the MikoPBX installation.

We recommend allocating at least 50 GB for this disk.
All data on the disk where MikoPBX is being installed will be lost.
Go to the "Program Code" tab:
Insert the following code:
Save the changes and copy the dialplan application identifier from the browser's address bar. It will look like "DIALPLAN-APP-CF967EE0".
Go to System → Customizing System Files and open the file /var/spool/cron/crontabs/root for editing.
Select the mode "Append to the end of the file", and in the black editing field at the bottom, insert the following code:
Save the changes.
You're done!
To enable notifications, you will need to configure the SMTP client. See instructions in the section "Mail and Notifications".


<?php
use MikoPBX\Common\Models\PbxSettings;
use MikoPBX\Common\Models\LanInterfaces;
use MikoPBX\Core\System\Notifications;
use MikoPBX\PBXCoreREST\Lib\SIPStackProcessor;
require_once 'Globals.php';
$result = SIPStackProcessor::callBack(['action' => 'getRegistry']);
$adminMail = null;
$localIP = '';
$message = '';
foreach ($result->data as $provider) {
if ($provider['state'] !== 'OFF' && $provider['state'] !== 'REGISTERED') {
if ($adminMail === null) {
$adminMail = PbxSettings::getValueByKey('SystemNotificationsEmail');
$interfaces = LanInterfaces::find("disabled='0'");
foreach ($interfaces as $interface) {
if (!empty($interface->ipaddr)) {
$localIP = $interface->ipaddr;
break;
}
}
}
$message .= "Provider state: {$provider['state']}<br>" . "Url (local): ']} <br><br>";
}
}
if (!empty($message) && !empty($adminMail)) {
$notify = new Notifications();
try {
$notify->sendMail($adminMail, 'Provider invalid state...', $message);
} catch (\Throwable $e) {
}
}*/5 * * * * /usr/bin/php -f /var/lib/asterisk/agi-bin/DIALPLAN-APP-9С060381.php start > /dev/null 2> /dev/null ssh -i C:\Users\<Username>\.ssh\id_ed25519 do-user@mikopbxipadressThe default login for SSH on a DigitalOcean VM is do-user.
Adjust:
C:\Users\<Username>\.ssh\id_ed25519 to the path of your local SSH key.
do-user if you changed the root user during VM creation.
mikopbxipadress to the IP address of your station (IPv4 in the VM management interface).













mkdir -p /storage/usbdisk1/transfercd /storage/usbdisk1/transfertouch transfer-rsync.shcurl -o /storage/usbdisk1/transfer/transfer-rsync.sh https://gist.githubusercontent.com/excla1mmm/c9891306b459cac0c7ea3c785ab0936e/raw/ec57ab60ee48112b4a16635e7b47955e5a044513/transfer-rsync.shchmod +x transfer-rsync.sh./transfer-rsync.shAfter the transfer, always verify the integrity of all data before retiring the old MikoPBX!








Installing MikoPBX using Proxmox.
Open the local / ISO images tab and select Download from URL
In the URL field, paste the link to the MikoPBX distribution file with the .iso extension
Click the Download button and wait for the file to finish downloading
Select Create VM
On the General tab, enter the name of the virtual machine, for example mikopbx-vm
Go to the OS tab, and in the ISO image field, select the previously downloaded image
Set the OS type to Linux
On the System tab, uncheck the Qemu Agent box, and use the default values for other fields
Go to the Disks tab
Adjust the size of the system disk to 1 GB
Click the Add button and add an additional disk for data storage
Specify a disk size of at least 50 GB
On the CPU and Memory tabs, specify the computing resources for the virtual machine based on the expected load on the PBX. For a test machine, you can set Cores (CPU tab) to 2 and Memory (Memory tab) to 2 GB
On the Network tab, uncheck the Firewall box
Go to the last tab, Confirm, and check the Start after created box
After entering the values, click the Finish button
Go to the created virtual machine mikopbx-vm
On the open tab, go to the Console section
If the boot is successful, a console menu will appear. Enter 8 from the keyboard to start the installation
Select the disk for the system and enter the disk name from the keyboard, for example sda. Confirm the selection by entering y from the keyboard
Connect the disk for storing call recordings, enter the disk name for connection from the keyboard, for example sdb
To access the MikoPBX web interface, enter your virtual machine's IP address in your browser's address bar. You can find the IP address in the console.
Enter the IP address in your browser’s address bar. Log in using the default credentials.
The QEMU Guest Agent lets Proxmox read the guest IP addresses from MikoPBX, perform a clean OS-level shutdown from the Shutdown button, and take consistent backups (fs-freeze). The qemu-ga binary is already shipped with MikoPBX and is automatically started by monit (the vm-tools service) — you only need to enable the agent channel on the Proxmox side.
Select the MikoPBX virtual machine, open the Options tab and double-click the QEMU Guest Agent row
Tick Use QEMU Guest Agent (set to Enabled), keep Type = VirtIO (default) and click OK
On the Proxmox host (quick alive/dead check):
Inside MikoPBX (via SSH):
If monit summary shows vm-tools Initializing and /dev/virtio-ports/org.qemu.guest_agent.0 is missing, the virtio-serial channel has not been attached to the VM yet — either the option is not enabled on the Proxmox side, or the VM has not been cold-restarted after enabling it.
Creating and configuring call queues.
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.
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 Agents 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 agents at the same time until someone answers the call (default behavior).
Linear: Calls are sent to agents one by one in the order configured in the Queue Agents list. After the Time attempt call to agents interval expires, the call is sent to the next agent.
Linear Progressive: The first agent starts ringing immediately. After each Time attempt call to agents interval, the next agent is added while the previous agents continue ringing. Make sure the timeout in Scenario 1 is long enough for all required agents to be added.
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.
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, the call will be redirected to the selected number. If no redirect destination is selected, the overall queue timeout is not applied.
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.
Installing MikoPBX using VMware Fusion.
Creating a new virtual machine.
After downloading the latest version of the image (link), specify the ISO file with the installation distribution.
Click "Continue"
Select the type of operating system Other Linux 5.x and later kernel 64-bit
Click "Continue"
Choosing the Legacy bios type
Click "Continue"
Click "Finish"
After creating a virtual machine, wait for it to load
Go to the section "[3] Reboot the system"
Choose "[2] Shutdown"
After shutting down the virtual machine, go to Settings
Select "Add device"
Select "New Hard Disk"
Click "Add..."
Choose the size of the hard drive (we recommend at least 50 GB)
Click "Apply"
Start the virtual machine
Select "[8] Install"
Enter the name of the disk on which MikoPBX will be installed
In our case - sdb, enter its name and press Enter
Confirm the disk selection: enter y
Select a disk for recording conversations
In our case - sdc, enter its name and press Enter
The system will reboot and MikoPBX will be ready for use.
The PBX displays the IP address of the station by which you can connect to it
Enter the IP address of the station in the browser bar and the MIkoPBX login menu will open:
Updating MikoPBX running in Docker or Docker Compose
We recommend updating sequentially, without skipping releases and versions.
A MikoPBX Docker container is not updated with .img or .iso files. To move to a new version, you need to download the new Docker image and recreate the container, preserving the /cf and /storage directories, network settings, and environment variables.
Create a backup of the MikoPBX settings. If the backup module is unavailable in the unprivileged container, prepare backup of the directories or volumes on the Docker host side.
Write down the name and version of the current image:
Save the current container configuration:
Check the /cf and /storage mounts:
Write down the ports in use, network_mode, hostname, and environment variables. If the container is started via Compose, save the current docker-compose.yml and the .env file.
End active calls and schedule a maintenance window.
For the latest stable version, run:
If a fixed tag is used, specify the same registry and the tag of the required version instead of latest.
Stop the container:
Create a consistent backup of /cf while the container is stopped. For the bind mount from the example above, you can run:
In addition to /cf, it is worth backing up the CDR database before the new version — it is located in /storage and may also be migrated on first start. /cf contains only mikopbx.db (the settings), while the call history is a separate cdr.db (and recording_storage.db) in /storage/usbdisk1/mikopbx/astlogs/asterisk/.
The backup must be made with the container stopped (as for /cf) so that the database is in a consistent state:
When rolling back, restore both archives: first /cf, then the CDR — otherwise the call history and the link between recordings and calls may get out of sync with the settings.
Rename the old container instead of deleting it immediately:
Start a new container with the same parameters. Example for an installation with host network and bind mounts:
Monitor the startup:
To exit the log view, press Ctrl+C — the container will keep running.
Go to the directory containing docker-compose.yml.
If user IDs are passed through the environment, set them the same way as on first launch:
Download the new image:
Recreate the container:
Check the status and logs:
Docker Compose will recreate the container but keep the bind mounts and named volumes specified in docker-compose.yml.
Make sure the container is running:
Open the web interface and check the MikoPBX version.
Check that the settings and call recordings are in place.
Make sure the phones and SIP providers have registered.
Make a test inbound and outbound call.
If the new container does not start:
Save its logs:
Stop and rename the new container:
Restore the /cf backup created before the update. The restoration method depends on whether a bind mount, a named volume, or a storage snapshot is used.
Give the old container its previous name back and start it:
To roll back a Docker Compose installation, specify the previous image tag in docker-compose.yml, restore the previous state of /cf, and run again:
After successful verification, you can delete the old container:
Description of section functions
The "System Diagnostics" section in MikoPBX is a tool for monitoring and analyzing the status of the telephone system. Using this section helps to promptly detect and eliminate technical problems, ensuring stable and efficient operation of MikoPBX.
Section location: "Maintenance" -> "System log entries"
Contains 3 tabs: Show log, System information, Capturing network packets.
This section allows you to view log files for detailed analysis of PBX operations.
To start, select a file.
Set the lines for the number of lines to fetch.
Set the offset value to shift the selection.
Set the filter by entering a string to be included in the selection.
The following options are available:
Download the selected log as a file .
Refresh the log .
Auto-refresh the log .
Suppose you need to analyze an outgoing call to the number 74952293042.
Select the log file asterisk/verbose.
Set the phone number 74952293042 as the filter.
Set the limit to a sufficient value, such as 2000, to ensure all log entries are included.
In the last line of the log selection, find the identifier:
In this example, the ID = C-0000000f.
Repeat the log query but use the filter C-0000000f this time.
You will receive the entire log of the dialplan process.
The obtained data can be sent to technical support for further assistance.
The tab displays the following information:
Network settings
CPU load
RAM usage
iptables settings
This section allows you to perform a detailed analysis of errors in PBX operation.
You can start capturing network packets passing through the network interface.
To start the process, press the Start button.
Reproduce your issue: make a call or perform an action that causes the error or failure.
Press the Stop and Download button.
The network packets will automatically be saved in the archive MikoPBXLogs_log-tcpdump-XXXXXXXXXX. They should also automatically save in your browser's Downloads folder.
If you cannot find the archive in the Downloads folder, you can and download it from the /storage/usbdisk1/mikopbx/tmp directory - the file will be log-tcpdump-XXXXXXXXXX.zip.
You can download all system logs accumulated on the PBX. To do this, click the Download All System Logs button.
The system logs will automatically be saved in the archive MikoPBXLogs_log-sys-XXXXXXXXXX.zip. The logs should also automatically save in your browser's Downloads folder. You can also obtain this log archive from the /storage/usbdisk1/mikopbx/tmp directory by connecting to the PBX using WinSCP. The file will be log-sys-XXXXXXXXXX.zip.
Other ways to collect logs:
Using an
Using the
Using the
MikoPBX Installation Guide using Docker container
To work with MikoPBX in a container, you need to install Docker and Docker Compose, as well as create a user and directories for storing configuration settings and call recordings according to the instructions
Docker installation and creating a user and directoriesTo launch the container with your application, use the following commands:
# Pulling the container image
sudo docker pull ghcr.io/mikopbx/mikopbx:latest
## Alternatively, you can download the image from Docker Hub.
# sudo docker pull mikopbx/mikopbx:latest
# Running the container in unprivileged mode
sudo docker run --net=host --name mikopbx --hostname mikopbx \
-v /var/spool/mikopbx/cf:/cf \
-v /var/spool/mikopbx/storage:/storage \
-e SSH_PORT=23 \
-e ID_WWW_USER="$(id -u www-user)" \
-e ID_WWW_GROUP="$(id -g www-user)" \
-it -d --restart always ghcr.io/mikopbx/mikopbx:latestTo ensure that your MikoPBX application is posted and working in the Docker container, you can follow these steps after launching it. These steps will help you verify the container's status and view its logs.
First, ensure that the container is successfully launched and running. To do this, use the command docker ps, which will show a list of running containers and their statuses.
sudo docker psThis command will display information about all active containers. Make sure that the mikopbx container is present in the list and its status indicates that it is running (e.g., status up).
After confirming that the container is running, the next step is to view the logs to ensure that the application has loaded without errors and is functioning properly. The docker logs command will allow you to see the output generated by your application.
Check the command output for a message similar to the one below. This message indicates that MikoPBX is successfully loaded and ready for use:
If you see the MikoPBX startup process, wait a moment and re-run the command sudo docker logs mikopbx
When the container starts, it lacks information about the host system's address, so you need to open the external address of the host system, in this case, Ubuntu, in a web browser. https://<host machine IP>
Log into the web interface using the admin login and the admin password to make sure that the web interface is accessible and functioning correctly.
If you need to use the "", the container should be run with the –privileged flag. When MikoPBX is run in a container, backups can also be performed by manually archiving the cf and storage directories. In this case, the privileged mode is not necessary, but the container must be stopped during copying.
The –net=host flag indicates that NAT between the host and container will not be used. MikoPBX will be directly connected to the host machine's network. All ports that the container needs to occupy will also be occupied on the host machine. If any port on the host machine is unavailable, errors will occur when loading MikoPBX. More details in the
If necessary, you can adjust the standard set of ports used by MikoPBX. This can be done by declaring environment variables when launching the container.
In addition to using our official registry, you might need to create a container from an image, for example, for a beta version. Our published releases and pre-releases include a tar archive, which we use to create a container.
Here is an example code for its use:
Do not forget to configure the external firewall correctly for your PBX. You can find the setup instructions .
Below are some of the environment variables that will allow you to adjust the MikoPBX ports and settings used.
SSH_PORT - port for SSH (22)
WEB_PORT - port for the web interface via HTTP protocol (80)
WEB_HTTPS_PORT - port for the web interface via HTTPS protocol (443)
SIP_PORT
A full list of all possible setting parameters is available in the source code .
Gmail service mail configuration via OAuth2 Authentication
Before starting the setup, you need to change some Google account parameters. To do this, go to the account management page (link).
Go to the "Security and sign-in" section. Make sure that two-step authentication is configured.
Go to the Google Cloud Console, to the "APIs & Services" section (). Create a project for the current task.
Go to the APIs library (the "Library" section).
Enter "gmail api" in the search bar. Open the Gmail API card.
Click "Enable" to connect.
Go to the main APIs & Services page. Then click "OAuth consent screen".
Create a project (click "Get started"). Fill in an arbitrary name and your email. Select "Internal" as the Audience. Click "Create" to finish.
Return to the main APIs & Services page. Then go to the "Credentials" section. Click "Create credentials". Select "OAuth client ID" to create.
Select "Web application" as the Application type. Then enter an arbitrary name. Click "Create".
Add a new "Authorized redirect URI".
An OAuth client will be created. Save the Client ID and Client Secret to your notes. You will need this data for the connection in the future.
Go to the "System" -> "Mail and notifications" section:
Next, go to "SMTP Settings". Fill in the following parameters:
Sender address, Sender name — your email and the name from which the emails will be sent.
Authentication type — OAuth2.
SMTP login — your email.
OAuth2 Provider — Google/Gmail.
Leave all other settings at their default values. A more detailed description can be found in the main article about mail parameters ().
After that, click "Save"!
Click the blue "Connect via OAuth2" button. Then select your Gmail account.
Confirm the sign-in: click "Continue".
Confirm granting the required permissions (click "Allow").
Upon successful authorization, you will see the following window.
Access blocked: Authorization Error (**Error 400: invalid_request)
Solution: enter the station's URL address in the MikoPBX web interface: "Network and Firewall" -> "Network Interfaces". Go to the "Network Topology" section and enter the hostname in the "External hostname of your router" field. (Enable "This station is located behind a NAT router".)
Use this guide when the disk that contains the local MikoPBX storage has been expanded in the hypervisor or on the physical server, but the /storage/usbdisk1 partition still shows the old size.
The local storage contains call recordings, call history, system logs, additional modules, backups, and system caches. In a standard installation, the storage is mounted at /storage/usbdisk1.
Before changing partitions, create a MikoPBX backup and a virtual machine snapshot or disk backup using your hypervisor tools.
You can run diagnostics over SSH. Run the actual resize operation from the local console, hypervisor console, or serial console. The standard resize script calls /sbin/freestorage, which stops the SSH server dropbear, so the SSH session will be disconnected.
Use this scenario if:
MikoPBX is installed on a virtual machine or a physical server;
the storage disk has already been expanded at the hypervisor, RAID controller, or storage subsystem level;
df -h /storage/usbdisk1 still shows the old file system size;
the disk has unallocated space after the current storage partition.
This guide does not apply to Docker installations. In Docker, the size of the /storage directory is managed by the host operating system and volume settings.
Connect to MikoPBX over SSH for diagnostics only, or open the MikoPBX console and run:
Find the disk listed in the m_Storage table. For example:
In this example, the storage disk is /dev/vdc, and the storage partition is /dev/vdc1.
Save the disk name to a variable:
Check the disk size and unallocated free space:
After the disk has been expanded in the hypervisor, the output must show that the disk itself is larger and that Free Space appears after the existing partition.
Example state before resizing:
If there is no free space, first expand the disk in the hypervisor or check that you selected the correct disk.
On GPT disks, after expanding a virtual disk, parted may show this warning:
In this case, first move the backup GPT header to the end of the new disk size:
Then repeat the check:
The Not all of the space... warning should disappear. The free space after the partition should still be visible.
Open the local MikoPBX console or the virtual machine console.
In the console menu, select:
Confirm stopping processes:
Enter y.
MikoPBX will run the standard scenario:
stop services that use storage;
unmount /storage/usbdisk1;
expand the storage partition to the end of the disk;
run e2fsck
If you need to run the same scenario as a command from the console, use:
After MikoPBX boots, identify the storage disk again and run the checks:
In a correct state:
/storage/usbdisk1 is mounted from the same storage partition;
the size of /storage/usbdisk1 has increased;
there is no free space after the storage partition, or only a minimal technical gap remains;
Example result after a successful resize:
Check that the data is accessible:
The correct response for the call history database is:
Check services:
Immediately after reboot, some monit checks may be in the Initializing state. Wait 1-2 minutes and repeat the check.
If Resize storage reported success but df -h /storage/usbdisk1 still shows the old size, check GPT and free space:
If the Not all of the space... warning is present, run:
Then run the resize operation again from the console:
If the free space is less than 5% of the disk size, the standard script may finish without resizing. In this case, check whether the disk was actually expanded enough.
If the system does not boot or storage is not mounted after resizing, use the backup or snapshot created before the operation.
If MikoPBX boots but storage is not mounted, check the UUID and database record:
If needed, connect the storage disk again using the standard script:
Description and configuration of outgoing routing
Outgoing routes in MikoPBX are a set of rules and settings that determine how the system handles outgoing calls from employees to external numbers. With their help, administrators can control the direction of calls through different telephony providers or communication lines depending on certain conditions, such as the dialed number, prefixes, time of day or user access rights. This allows you to optimize communication costs, distribute the load between channels and apply security policies by restricting or allowing certain types of calls. Setting up outgoing routes provides flexibility and control over outgoing telephone communications, contributing to the efficient operation of the company's communication system.
In this article, you will find detailed documentation on setting up outgoing routing.
To add a new outgoing routing rule, click the "Add a new rule" button.
The name of the rule can be set arbitrarily.
In a note, you can describe the call route that you want to implement; this can help you in debugging in the future.
Set a template for outgoing calls. Read more about number templates in this .
The example in the picture above means the following: if the dialed number starts with "345" or "375" and the rest of the number consists of 10 digits.
Convert number - this setting is intended to remove the number prefix and replace it with the desired prefix.
Set a template for outgoing calls. Read more about number templates in the .
In the example given, digits are not cut off at the beginning of the number and digits are not added.
Select from the list the provider for which you configured outgoing routing and save the changes.
For example, the number +74952293042 should be converted to the number 84952293042.
The implementation of the rule looks like this:
For example, the numbers 84952293042 and 74952293042 should be converted to +74952293042.
The implementation of the rule looks like this:
For example, the numbers 4952293042 and 4996382584 should be converted to 84952293042 and 84996382584 respectively.
The implementation of the rule looks like this:
For example, the numbers 84952293042 and 74996382584 should be converted to 2293042 and 6382584 respectively.
The implementation of the rule looks like this:
Description of the registration process
Registering in the MikoPBX Module Marketplace is not required for the system’s basic functionality. You can fully utilize MikoPBX for handling calls without registration or installing additional modules. However, we recommend registering in the marketplace to expand your system’s capabilities.
Registration gives you access to additional modules and extensions. These include both free modules (moved out of the core system for easier initial setup) and paid modules from us and third-party developers. If you are a developer, you can contact us at developers@mikopbx.com to get instructions on creating and adding your module to the Marketplace.
MikoPBX is a free solution and does not require registration. The absence of a license does not impact call functionality. You can register or cancel your Marketplace registration anytime.
To begin the registration process, navigate to "Modules" -> "Marketplace of modules":
If you are not registered in the Marketplace, the section will look as follows:
If you move MikoPBX to another host or restore it from a backup, you will need to reset license bindings in the .
To start the registration process, click the Register in Marketplace button:
The license key stores all your licenses for MikoPBX products. If you already have a key, you can enter it in the corresponding field. If you've forgotten your key, search your email inbox for messages from lic@miko.ru
If you don’t have a key, you can generate a new one by completing the registration form:
Organization Name – Enter your company/organization name.
Contact Email – Enter your organization’s email address.
Contact Person – Enter the name of the contact person.
Contact Phone (optional) – Provide a contact number.
Click Register.
Upon successful registration, you will see the following screen:
A notification confirming the system's registration.
The license key field will display a blurred value by default. Hover over it to view or copy the key.
To manage your license, go to Marketplace -> License Management and click the corresponding option:
You will be redirected to the :
Enter your license key in the Enter your license key or activated coupon field and click Login:
You will access a system with nine sections:
Go to the Session monitor section:
In the Info column, click the i button for each binding to view detailed host information.
In the Action column, use the Drop button to unbind the license from the current host.
If you encounter issues during registration, check for internet access to the MikoPBX server. Ensure connectivity to lic.miko.ru and lic.mikopbx.com over port 443 (HTTPS). Verify firewall settings and network permissions.
MikoPBX periodically connects to licensing servers to verify installed modules. If a module license becomes unavailable, the module will be disabled, and a strikethrough key icon will appear next to its name.
Description of usage in MikoPBX
REST API MikoPBX allows you to automate station management and integrate it with external systems - CRM, helpdesk, corporate portals, and custom services. API keys are used to access the API.
All REST API requests are authorized via the Authorization: Bearer <token> header. MikoPBX supports two token types:
JWT token
For external integrations, always use an API key — it is created manually, has configurable access permissions, and can be revoked at any time.
Go to "System" → "API Keys".
`Click "Add API Key".
Fill in the Description field (e.g.: CRM Integration)
Copy the generated API key — it is displayed only once
Important: save the key immediately after creation. Once the page is closed, it cannot be recovered — you will need to create a new one.
Follow the principle of least privilege — each key should only have access to the resources that are actually needed.
When creating a key, two options are available:
Full access — the key gets read and write access to all API resources. Use only if truly necessary.
Manual configuration — the access level for each API resource is specified individually: read-only, read and write, or no access.
Network filter: choose one of two options:
Localhost connections only — the key will only work from the local network. Recommended if the integration runs within the infrastructure.
Connections from any address are allowed — the key is accessible without IP restrictions. Use only if the client is located outside the local network.
Following these requirements protects the API from token interception and unauthorized access:
Valid SSL certificate:
Use a trusted SSL certificate on the MikoPBX server side. The easiest way is to issue a free certificate via the Let's Encrypt module (instructions for working with the module are available here).
Operating without a valid certificate is only acceptable in an isolated test environment with no internet access.
Certificate trust on the client side:
The client must verify the server certificate on every request. Disabling verification (verify=False in Python, -k in curl) is not acceptable in production: without it, a man-in-the-middle (MITM) attack becomes possible, where an attacker intercepts the Bearer token in plaintext.
Key scope restriction:
Each key must only have access to the resources actually used by the integration. Do not use "Full access" unless necessary — compromising such a key gives an attacker full control over the API.
Network access restriction:
If the integration runs within a local network — choose "Local connections only". This prevents a compromised key from being used from an external network.
Use "Allow connections from any address" only when the client is physically located outside the local network, and make sure all other security measures are in place — a valid SSL certificate and minimal key permissions.
Click a card to navigate:






















































Installing MikoPBX as a guest machine in VirtualBOX
Download Virtual Box from the and install it.
Create a new virtual machine.
Specify the
Installing MikoPBX using the Vultr cloud platform
This guide provides a step-by-step process for installing MikoPBX on the Vultr cloud platform.
Before starting, you must download the latest .iso MikoPBX image file from .
First, you need to upload the image to the cloud platform.
Navigate to "Cloud Storage" → "Object Storage":
Instructions for connecting Wasabi Cloud Storage as an S3 storage
Go to the Wasabi console ().
In the left menu, select "Buckets" and click "Create Bucket".
On the bucket creation page, specify:
Mail setup for the Outlook service (outlook.com; hotmail.com) via OAuth2 Authentication
Sign in to the
Go to "Entra ID" -> "App registrations". Then click "New registration" to register a new application.
Description of the capabilities of the "System file customization" section
The system file customization section can be found under "System" -> "System file customization":
This section allows for customization of system and Asterisk configuration files. We recommend using this section only for experienced Asterisk administrators. MikoPBX provides the ability to modify the following configuration files via the web interface:
Instructions for connecting AWS S3 as cloud storage for automatic uploading of call recordings from MikoPBX
Go to the AWS console (). Navigate to "All services" -> "Storage" -> "S3".
Click "Create bucket".
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 answered calls within the queue.
Random: A random available agent within the queue is selected to receive the call.
Memory Hunt: The system remembers the last agent who answered a call and starts the next distribution from the following agent.
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.
In Scenario 1, specify both the waiting time and the redirect destination. If only the time is filled in without a backup destination, MikoPBX does not pass the overall timeout to the queue application so that the call is not ended without a route to continue.









Internal system components, modules, built-in tools
API key
External integrations: CRM, scripts, third-party services

































Allows the user to set information on channels
Log
Provides the user with access to reading logs
Read only
Verbose
Provides the user with access to reading detailed logs
Read only
Agent
Reading agent status events from app_queue and chan_agent modules
Allows the user to perform actions to manage and retrieve the status of queues and agents
User
Access to user events as well as Jabber/XMPP user events
Allows the user to execute the UserEvent command to create custom events
Config
For recording only
Allows the user to receive, update, and overload configuration files
Command
For recording only
Allows the user to execute Asterisk CLI commands from AMI
DTMF
Allows the user to receive DTMF events
Read only
Reporting
Access to call quality events such as jitterbuffer or RTCP
Allows the user to perform a number of actions to obtain statistics and information about the status of the entire system
Cdr
Reading data write events in CDR
Read only
Dialplan
Reading events for setting dialplan variables, creating extents
Read only
Originate
For recording only
Allowing the user to execute the Originate command, which sends a request to create a new call




































Application ID (Client ID), Secret key (Client Secret) — the data saved from Google Cloud (step 12 from the previous section of this guide).




















Unique Company Identifier (e.g., Tax ID, VAT) (optional).













79257184255, 7925, 7925718…
7ХХ
0
700, 701, 702…
74952293042
0
74952293042
74(95|99)
7
74952293042, 74996382584…
(7|8)0{1}
1
700, 701, 802, 803…
(25|26)
0
25, 26
[0-9]{1}
0
digit from 0 to 9, occurrence once
[1-5]{2}
0
12, 15, 14, 25 digit from 1 to 5, occurrence twice
[8-9]+
0
8899, 888, 988888 digit from 8 to 9, occurrence one or more times
[7-8]{1}
10
79257184255, 84952293042
7925
If the dialed number matches the rules of several routes, then the call will be made in the order of the route descriptions, one by one, until the call is answered, or until there are no more suitable routes.










leave the field blank
Check the publishing of web, SIP, and RTP ports if a bridge network is used.
The example must not be copied without verification. Repeat all the parameters of your exact previous installation: network type, volumes, ports, hostname, capabilities, and environment variables.
On first start, the new version may have upgraded the database in /cf. Therefore, a reliable rollback must rely not only on the old container image but also on the /cf backup created before the update.
Be careful! If there are many calls or heavy network "load" on the PBX, logs can take up a significant amount of disk space.










run resize2fs;
reboot the system.
/etc/fstab matches the UUID of the storage partition;the m_Storage table points to the same storage disk.
Do not run this command from an SSH session. During execution, dropbear will be stopped and the connection will be disconnected.
sudo docker inspect mikopbx --format '{{.Config.Image}}'sudo docker inspect mikopbx > mikopbx-container-before-update.jsonsudo docker inspect mikopbx --format '{{range .Mounts}}{{println .Source "->" .Destination}}{{end}}'sudo docker pull ghcr.io/mikopbx/mikopbx:latestsudo docker stop mikopbxsudo tar -C /var/spool/mikopbx -czf mikopbx-cf-before-update.tar.gz cfsudo docker stop mikopbx
# backup of /cf (settings)
sudo tar -C /var/spool/mikopbx -czf mikopbx-cf-before-update.tar.gz cf
# backup of the CDR + recording metadata
sudo tar -C /var/spool/mikopbx/storage/usbdisk1/mikopbx \
-czf mikopbx-cdr-before-update.tar.gz astlogs/asterisksudo docker rename mikopbx mikopbx-before-updatesudo docker run --net=host --name mikopbx --hostname mikopbx \
-v /var/spool/mikopbx/cf:/cf \
-v /var/spool/mikopbx/storage:/storage \
-e SSH_PORT=23 \
-e ID_WWW_USER="$(id -u www-user)" \
-e ID_WWW_GROUP="$(id -g www-user)" \
-it -d --restart always ghcr.io/mikopbx/mikopbx:latestsudo docker logs -f mikopbxexport ID_WWW_USER=$(id -u www-user)
export ID_WWW_GROUP=$(id -g www-user)sudo docker compose pullsudo docker compose up -dsudo docker compose ps
sudo docker compose logs -f mikopbxsudo docker ps --filter name=mikopbxsudo docker logs mikopbx > mikopbx-update-error.log 2>&1sudo docker stop mikopbx
sudo docker rename mikopbx mikopbx-failed-updatesudo docker rename mikopbx-before-update mikopbx
sudo docker start mikopbxsudo docker compose up -dsudo docker rm mikopbx-before-update[2023-03-18 16:28:17] VERBOSE[21654][C-0000000f] app_stack.c: Spawn extension (SIP-1676196499-22-outgoing, 74952293042, 13) exited non-zero on 'PJSIP/302-0000001a'lsblk -f
df -h /storage/usbdisk1
mount | grep /storage/usbdisk1
cat /etc/fstab
sqlite3 /cf/conf/mikopbx.db "select id,name,device,uniqid,filesystemtype,media from m_Storage;"1|Storage №1|/dev/vdc|854d0b18-608a-4634-b910-dea3726ae1a5|ext4|1STORAGE_DISK=$(sqlite3 /cf/conf/mikopbx.db "select device from m_Storage where media='1' limit 1;")
echo "$STORAGE_DISK"blockdev --getsize64 "$STORAGE_DISK"
parted -s "$STORAGE_DISK" unit MiB print freeDisk /dev/vdc: 12288MiB
Number Start End Size File system Name Flags
0.02MiB 1.00MiB 0.98MiB Free Space
1 1.00MiB 10239MiB 10238MiB ext4 primary
10239MiB 12288MiB 2049MiB Free SpaceWarning: Not all of the space available to /dev/vdc appears to be usedsgdisk -e "$STORAGE_DISK"
sync
blockdev --rereadpt "$STORAGE_DISK" 2>/dev/null || true
partprobe "$STORAGE_DISK" 2>/dev/null || trueparted -s "$STORAGE_DISK" unit MiB print freeStorage -> Resize storageAll processes will be completed. Continue? (y/n):/etc/rc/resize_storage_part "$STORAGE_DISK"STORAGE_DISK=$(sqlite3 /cf/conf/mikopbx.db "select device from m_Storage where media='1' limit 1;")
echo "$STORAGE_DISK"
lsblk -f
df -h /storage/usbdisk1
parted -s "$STORAGE_DISK" unit MiB print free
mount | grep /storage/usbdisk1
cat /etc/fstab
sqlite3 /cf/conf/mikopbx.db "select id,name,device,uniqid,filesystemtype,media from m_Storage;"Filesystem Size Used Available Use% Mounted on
/dev/vdc1 11.7G 1.1G 10.0G 10% /storage/usbdisk1test -d /storage/usbdisk1/mikopbx && echo "mikopbx directory OK"
test -f /storage/usbdisk1/mikopbx/astlogs/asterisk/cdr.db && echo "cdr.db OK"
sqlite3 /storage/usbdisk1/mikopbx/astlogs/asterisk/cdr.db "pragma integrity_check;"okmonit summary
pbx-console services status
asterisk -rx "core show uptime"STORAGE_DISK=$(sqlite3 /cf/conf/mikopbx.db "select device from m_Storage where media='1' limit 1;")
parted -s "$STORAGE_DISK" unit MiB print freesgdisk -e "$STORAGE_DISK"
sync
blockdev --rereadpt "$STORAGE_DISK" 2>/dev/null || true
partprobe "$STORAGE_DISK" 2>/dev/null || true/etc/rc/resize_storage_part "$STORAGE_DISK"blkid
cat /etc/fstab
sqlite3 /cf/conf/mikopbx.db "select id,name,device,uniqid,filesystemtype,media from m_Storage;"
mount | grep /storage/usbdisk1printf "%s\n" "$(basename "$STORAGE_DISK")" | /etc/rc/connect_storageTLS_PORT - port for connecting a SIP client with encryption (5061)
RTP_PORT_FROM - beginning of the RTP port range, voice transmission (10000)
RTP_PORT_TO - end of the RTP port range, voice transmission (10800)
IAX_PORT - port for connecting IAX clients (4569)
AMI_PORT - AMI port (5038)
AJAM_PORT - AJAM port used for connecting the telephony panel for 1C (8088)
AJAM_PORT_TLS - AJAM port used for connecting the telephony panel for 1C (8089)
BEANSTALK_PORT - port for the Beanstalkd queue server (4229)
REDIS_PORT - port for the Redis server (6379)
GNATS_PORT - port for the gnatsd server (4223)
ID_WWW_USER - identifier for www-user (can be set with the expression
$(id -u www-user), where www-user is NOT a root user)
ID_WWW_GROUP - group identifier for www-user (can be set with the expression
$(id -g www-user), where www-user is NOT a root group)
WEB_ADMIN_LOGIN - login for Web interface access
WEB_ADMIN_PASSWORD - password for Web interface access
sudo docker logs mikopbx++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
| All services are fully loaded welcome |
| MikoPBX 2026.1.223 |
| built on Tue Apr 7 03:39:14 UTC 2026 (arm64) in Docker |
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
| Web Interface Access |
| |
| Local Network Address: |
| https://192.168.65.3 | |
| |
| Web credentials: |
| Login: admin |
| Password: admin |
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
| SSH access disabled! |
| |
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++# Create a container from a tar archive (you need to download it first!)
sudo docker load -i mikopbx-2026.1.223-x86_64-docker.tar
# Launch the created container
sudo docker run --net=host --name mikopbx --hostname mikopbx \
-v mikopbx_cf:/cf \
-v mikopbx_storage:/storage \
-e SSH_PORT=23 \
-e ID_WWW_USER="$(id -u www-user)" \
-e ID_WWW_GROUP="$(id -g www-user)" \
-it mikopbx:2026.1.223
Type - Linux.
Version - Other Linux (64Bit).
Click Next.
Specify the size of the base memory - 1024 MB, as well as the number of processors - 2
Press Next.
Select Create a new virtual hard disk. Enter a disk size of 700 MB, and also check the box "Pre-allocate Full Size"
Click Create.
Confirm the creation of the virtual machine: click Finish.
Go to the settings of the created virtual machine.
To do this, click "Settings" in the upper menu.
Click the "Storage" tab. Add a new hard drive to store call records.
In the window that appears, click Create.
Select the hard disk format - VDI (VirtualBox Disk Image).
Click Next.
The hard disk must be of a fixed size.
Check the box next to "Pre-allocate Full Size"
Click Next.
Specify the Name of the created disk.
Set the Disk Size to about 50 GB.
Click Finish.
Choose the newly created drive and click Select.
The created drive will appear in the media list.
Please mount the MikoPBX image onto an optical disc. To do this, select the optical disc in the 'Media' section and click on the image file selection button in the 'Attributes' section.
In the appeared menu, click on 'Choose a disk file...'
Select the downloaded ISO disk image.
"Go to the 'Network' tab.
Set the Connection Type to 'Bridged Adapter'. Click 'OK' to save all the settings you have made.
Start the created virtual machine.
The command interface of the PBX will open. The PBX will start booting.
At this stage, MikoPBX is booting from the optical disc containing the ISO image. This is indicated by the message: 'The system is loaded in Recovery mode'.
Install MikoPBX.
Click Install.
Information about all available disks will be displayed (in this example: sdb, sdc).
Enter the name of the disk you referred to as the 'system disk' from the keyboard, in this case sdb, and press Enter. (If it is selected by default, you can simply press Enter).
The system will prompt for confirmation. Enter 'y' and press Enter.
After completing the installation, you will be prompted to select a disk for storing call records.
Enter the disk name (in this example, the only available disk is 'cdc') and press Enter.
After the installation is complete, the system will reboot.
MikoPBX will now run from the sdb drive where you installed it.
We will see that the line "The system is loaded in Recovery mode" is missing.
To access the control panel, you need to enter the IP address of your virtual machine in the browser's address bar.
The installation of MikoPBX using VirtualBOX is now complete.
Use versions of MikoPBX below 2024.1.114 for installation on VirtualBOX
Version 2024.1.114 temporarily does not support installation on VirtualBOX

All data on the disk where MikoPBX is being installed will be lost
The disk where MikoPBX will be installed is referred to as the system disk (SYSTEM). You can choose a disk with a size larger than 500MB as the system disk.
Select the type of storage (it’s recommended to use the basic option, as you only need it to store the disk image). Also provide a name.
Click on your newly created storage resource:
Go to the "Buckets" tab and create a new bucket with a custom name.
The storage information will display S3 connection details.
Next, connect to your storage via WinSCP. Open WinSCP and select "New Site":
Enter the following parameters:
File protocol – Amazon S3
Encryption – TLS/SSL Implicit encryption
Port number – 443
Host Name, Access key ID, and Secret access key – from the storage information
Click "Login".
Upload the .iso disk image file to the storage.
Return to the Vultr interface and go to your bucket’s directory.
Click the three dots to the right of the file name, then "Change Access". Grant access by toggling the switch.
Click the three dots to the right of the file name and select "Copy URL".
Navigate to "Orchestration" → "ISOs":
Click "Add ISO":
Paste the link to your previously uploaded file and click "Upload".
Go to "Account" → "SSH Keys". Click "Add SSH Key":
Generate an SSH key pair following this guide.
In the interface for adding the key pair, provide a custom name and paste your public SSH key.
Click "Add SSH Key".
Go to "Products" → "Compute":
Click "Deploy Server":
In the next section, select the region and configuration for your virtual machine.
Continue configuring the server:
Under ISO/iPXE, select the previously uploaded ISO.
Select the SSH key pair you created.
Click "Deploy".
After the server is created, power it off.
Go to "Cloud Storage" → "Block Storage":
Click "Add Block Storage":
Select the disk type, region (same as the VM), size, and a custom name.
Go to the management page for the newly created block storage. Attach the volume to your virtual machine using the "Attach to:" option.
Go to your virtual machine management page.
Open the console by clicking the relevant button:
You will enter the built-in console.
Navigate to "[8] Install".
Select the disk to be used as the system disk. Confirm by typing "y" and pressing "Enter":
Select the disk for storing call recordings. The system will reboot.
Go to "Settings" for your virtual machine and then "Custom ISO". Click "Remove ISO".
At this point, MikoPBX is installed and ready to use.
In your browser, enter the IP adress of your virtual machine. You can find it in the MikoPBX console.
Log in using the following credentials:
Username: admin
Password: The VM ID, which looks like "150dd137-a0e2-45f6-baf9-ddc15a600d60" and can be found in the address bar (screenshot below).
This guide applies to MikoPBX version 2024.2.138 and later!

We recommend at least 50GB for storing call recordings. This guide uses 30GB as an example.
Bucket Name — enter any unique name for the bucket (e.g., mikopbx-s3-storage).
Region — select the region closest to your MikoPBX server.
Click "Create Bucket".
After creating the bucket, you need to create an access policy. Go to "Policies" in the left menu and click "Create Policy".
Enter a name for the policy (Policy Name) and a description for future identification (Description). In the "Policy Editor" field, paste the following set of rules:
Go to "Users" in the left menu (under "Users & Groups") and click "Create User".
On the first step "Details", fill in the following parameters:
UserName — enter any username (e.g., mikopbx-user).
Type of Access — check only "Programmatic (create API keys)".
Require MFA — leave disabled.
Click "Next".
On the Groups step — skip it and click "Next".
On the Policies step — select the policy you created earlier (e.g., mikopbx-access in this guide) and click "Next".
On the Review step, verify the parameters and click "Create User".
After the user is created, the Access Key and Secret Key will be displayed. Save these values — you will need them to configure MikoPBX. The Secret Key is shown only once.
Go to the "Maintenance" tab → "Storage".
Switch to the "S3 Cloud Storage" tab and fill in the following fields:
Automatically upload recordings to cloud storage — enable the toggle.
S3 endpoint URL — enter the endpoint for your region from the table below.
For example, for region eu-central-1: https://s3.eu-central-1.wasabisys.com
S3 region — specify the region of your Wasabi bucket (e.g., eu-central-1).
S3 bucket Name — specify the name of the bucket created in Wasabi (e.g., mikopbx-s3-storage).
Access Key and Secret Key — paste the values obtained when creating the Access Key.
Configure the "Local storage (S3 mode)" slider — select how long recordings will be stored locally before being deleted after upload to the cloud.
Click "Save".
us-east-1 (N. Virginia)
https://s3.wasabisys.com
us-east-2 (N. Virginia)
https://s3.us-east-2.wasabisys.com
us-west-1 (Oregon)
After saving the settings, click "Test Connection". If the connection is successful, the message "S3 connection successful" will appear and synchronization of call recordings will begin.

{
"Version": "2012-10-17",
"Statement": [
{
"Effect": "Allow",
"Action": [
"s3:PutObject",
"s3:GetObject",
"s3:DeleteObject",
"s3:ListBucket"
],
"Resource": [
"arn:aws:s3:::YOUR-BUCKET-NAME",
"arn:aws:s3:::YOUR-BUCKET-NAME/*"
]
}
]
}Name - enter a name for your application.
Supported account types - select "Accounts in any organizational directory (Any Microsoft Entra ID tenant - Multitenant)".
Specify the Redirect URL:
Select a platform — select "Web".
URL:
Replace 192.168.100.71 with your MikoPBX address.
Then click "Register".
The application will be created. Save the Client ID — you will need it in the future for configuration inside the MikoPBX web interface.
From the application's main page, go to "Manage" -> "API permissions".
Click "Add a permission".
In the "Microsoft Graph" section, select "Delegated Permissions". Enter "SMTP" in the search bar. Check the box next to "SMTP.Send".
Also enter "offline" in the search bar. Check the box next to "offline_access".
Click "Add permissions".
Next, go to "Certificates & secrets" -> "Client secrets". Click "New client secret".
Set the required parameters:
Description - an arbitrary description.
Expires - the duration for which you are issuing this client secret. It will be needed for application authentication in MikoPBX.
Click "Add".
Copy the "Value" (not the Secret ID!). It will be needed for configuration in the MikoPBX web interface.
For the application to work correctly, you need to grant permission to use the SMTP protocol for the user whose mailbox you are authorizing during this setup. To do this, follow these steps:
Go to the organization's admin center (link).
Go to "Users" -> "Active Users". Click on the name of the user account under which the application is being created.
In the account, go to the "Mail" section and select "Manage email apps".
Make sure that "Authenticated SMTP" is allowed. Save the changes by clicking "Save changes".
Go to the MikoPBX web interface. Then "System" -> "Mail and Notifications" -> "SMTP Settings".
Fill in all the required fields:
Sender address, Sender name — your email and the name from which the emails will be sent.
Authentication type — OAuth2.
SMTP login — your email.
OAuth2 Provider — Microsoft/Outlook.
Application ID (Client ID), Secret key (Client Secret) — data from Microsoft Entra.
Leave all other settings at their default values. A more detailed description can be found in the main article about mail parameters (link).
After that, click "Save"!
Click "Connect via OAuth2". Sign in to your Microsoft account. Then confirm granting all requested permissions.
Upon successful authorization, you will see the corresponding window.


https://192.168.100.71/pbxcore/api/v3/mail-settings/oauth2-callbackAfter creation, the Client Secret value will be shown only once. Do not forget to copy it into the MikoPBX web interface.
asterisk.conf
General (global) settings of Asterisk. In the asterisk.conf configuration file, you define the following: - The location, permissions, and owner of the socket used to connect the remote management console to the server. The location of various directories used by the Asterisk server to determine where configuration files, libraries, scripts, and logs will be created. Default command-line parameters for starting the server.
cel.conf
Channel Event Logging. Unlike CDR, it logs all events that occur in the channel.
extensions.conf
The Asterisk dialplan. It defines how incoming and outgoing calls are handled and routed. This file controls the behavior of all connections passing through your PBX.
features.conf
The file defines custom codes and options for Asterisk functions like call transfer, call pickup, on-demand recording, digit timeout, call parking, etc.
http.conf
Built-in Asterisk HTTP server configuration.
iax.conf
Describes your IAX protocol connections.
indications.conf
Nationalization of tonal signals.
logger.conf
Asterisk logging configuration. This file configures logging of Asterisk events to files, console, and Linux syslog. To apply settings, run the command "logger reload" in the Asterisk console (CLI).
manager.conf
AMI (Asterisk Manager Interface) configuration.
modules.conf
Asterisk module loading parameters.
musiconhold.conf
Music-on-hold settings in IVR.
queues.conf
Asterisk queue settings. Detailed description of call strategies, penalty, timeout, member, and other available parameters.
rtp.conf
Global RTP settings – media ports and protocol.
sip.conf
Configures internal and external SIP channels in Asterisk.
voicemail.conf
Email notification settings.
jail.local
Fail2ban settings.
msmtp.conf
SMTP client settings.
To edit a configuration file, use the button:
You will be presented with the configuration file editing form:
Choose one of four editing options:
"Not change" - prevents modifications, read-only mode.
"Add to end of file" - appends content to the end of the selected configuration file (recommended).
"To replace all" - your changes will completely overwrite the configuration file.
"Script" mode in MikoPBX system file customization allows administrators to add custom scripts or commands directly into the configuration files. This mode is ideal for advanced users who need to execute specific actions, automate tasks, or modify system behavior dynamically, enhancing the flexibility of the PBX configuration. It should be used with caution to avoid system disruptions.
In some cases, more complex modifications to system files are required than simply adding text to the end of a configuration file. For instance, you may need to redistribute PJSIP account parameters while retaining the ability to configure the system through the web interface.
We've introduced a new approach to customization, where you can describe a Bash script that will execute each time the system generates a configuration file. This way, integrators can make precise changes to configuration files without developing additional modules.
For example, you can modify the pjsip.conf file and change the max_contacts parameter for all internal numbers, except one.
You can add parameters to an existing section using the (+) syntax:
It is possible to intercept the execution of the dialplan by defining custom contexts. Currently, you can intercept executions in the following contexts:
internal-originate-custom - used for calls originating from the telephony panel for 1C.
<PROVIDER-ID>-incoming-custom - used for handling incoming calls from the provider.
<PROVIDER-ID>-outgoing-custom - used for handling outgoing calls via the provider.
all_peers-custom - used for direct outgoing calls from a phone.
outgoing-custom - used when dialing an external number, before selecting an outbound route.
add-trim-prefix-clid-custom - used for handling incoming calls, best suited for normalizing incoming phone numbers by adding/removing a prefix.
internal-users-custom - used for handling calls to internal extensions.
public-direct-dial-custom - used for handling incoming calls without authentication.
Example context:
Make sure to call the method "return" at the end.
File Name
Description

[user2_pingtel]
type=friend
username=user2_pingtel
secret=blah
host=dynamic
qualify=1000 ; Consider the client unreachable if response time exceeds 1 sec.
callgroup=1,3-4 ; The client is a member of call groups: 1, 3, and 4.
pickupgroup=1,3-4 ; We can "pick up" calls using *8 for calls in groups 1, 3, and 4.
defaultip=192.168.0.60
disallow=all
allow=ulaw
allow=alaw
allow=g729[outgoing-custom]
exten => _X!,1,NoOp(--- hangup - ${CHANNEL} ---)
same => n,returnModify the dialplan with caution – there is a high chance of disrupting the PBX!
Enter any name for the bucket (field "Bucket name"). Leave all other parameters as default and click "Create bucket".
Go to "All services" -> "Security, Identity, & Compliance" -> "IAM".
Next, create a new IAM user. Go to the "Access Management" tab, then "Users". Click "Create user".
Enter the name of the IAM user in the "User name" field.
Click "Next".
Select "Attach policies directly" as the "Permissions options". Scroll down the page.
In the "Permissions policies" section click "Create policy".
In the newly opened tab, in the "Policy editor", select "JSON" as the format and paste the following content into the parameters field:
Click "Next".
Next, specify any name for the policy being created.
Click "Next".
Return to the user creation tab, refresh the policy list, and select the previously created policy (in this guide — "access-mikopbx").
Click "Next".
Confirm user creation: click "Create user".
Open the page of the created user by clicking on the username.
Go to the "Security credentials" section. Click "Create access key".
Select "Application running outside AWS". Click "Next".
Enter a description for the key so that you can identify it later. Click "Create access key".
The Access key and Secret access key will be displayed. Save them — they will be needed later when configuring MikoPBX.
Go to the "Maintenance" -> "Storage" tab.
Open the "S3 Cloud Storage" tab and fill in the following fields:
Automatic recording upload to cloud storage — enable the switch.
S3 endpoint URL — enter the S3 AWS endpoint depending on the region of your bucket (link to the table with all URLs). In this guide — https://s3.ap-southeast-1.amazonaws.com
S3 region — specify the region of your bucket. In this guide — ap-southeast-1
S3 bucket name — enter the name of the bucket created in AWS (for example aws-s3-mikopbxstorage in this guide)
Access key and Secret key — paste the values obtained when creating the service account access key.
Configure the “Local storage period (S3 mode)” slider — choose how long recordings will be stored locally before being deleted after uploading to the cloud.
Click “Save”.
After saving the settings, click "Test connection". If the connection is successful, the message “S3 connection successful” will appear and synchronization of call recordings will begin.


{
"Version": "2012-10-17",
"Statement": [
{
"Effect": "Allow",
"Action": [
"s3:PutObject",
"s3:GetObject",
"s3:DeleteObject",
"s3:ListBucket"
],
"Resource": [
"arn:aws:s3:::your-bucket-name",
"arn:aws:s3:::your-bucket-name/*"
]
}
]
}Replace "your-bucket-name" with the name of the bucket you created earlier (in this guide — "aws-s3-mikopbxstorage").
The secret access key can only be viewed or downloaded now. It cannot be recovered later.
qm agent <VMID> ping # should return an empty response with no error
qm agent <VMID> info # agent version and supported commands
qm guest cmd <VMID> get-host-name # guest hostname — proves the agent actually replies
qm agent <VMID> network-get-interfaces # guest IP addresses (also visible in the VM summary)ls -la /dev/virtio-ports/org.qemu.guest_agent.0 # this file must exist
ps w | grep qemu-ga | grep -v grep # the qemu-ga process must be running
monit summary | grep vm-tools # expected state: RunningTo deploy the PBX use two disks:
A 1 Gb disk for the main system
A 50+ Gb disk for storing call recordings
When the message "Press any key within 30 seconds to boot from LiveCD..." appears, do not press any buttons. In this case, the system will boot from the hard drive.
Use the following default credentials for the first login to the MikoPBX web interface:
Username: admin
Password: admin
After enabling the option you must perform a full stop and start of the VM (Stop → Start in the WEB UI, or from the Proxmox CLI: qm stop <VMID> && qm start <VMID>).
A regular Reboot is not enough — the virtio-serial device that the agent talks through is attached to the QEMU process only on a cold start. Without that restart the /dev/virtio-ports/org.qemu.guest_agent.0 device never appears inside MikoPBX and the agent cannot start.






































Installing MikoPBX using the Alibaba Cloud platform
This guide applies to MikoPBX version 2024.2.135 and later!
This step-by-step guide will walk you through installing MikoPBX on the Alibaba Cloud platform.
Before starting, download the latest .raw MikoPBX image from MikoPBX’s GitHub releases.
First, create a bucket for storing your image. Go to the OSS Management Console (link).
Go to Buckets.
Click Create Bucket:
Specify the following:
Bucket name – a custom name for your storage.
Region – pick the region where your image will be stored.
Click OK.
Go to your newly created bucket by clicking its name in the Buckets section:
Click Upload object and upload the previously downloaded .raw disk image file (leave other parameters at default).
Once the disk image file is uploaded, copy its link. Click View Details to the right of the file name; in the opened menu, copy the URL field.
Return to the ECS Console () and go to Images.
Click Import Image:
Select Linux Operating System and click Next.
Enter/select the following image parameters:
Image File URL – Paste the link to the disk image file you uploaded.
Image Name – A custom,
Click OK to create the image. Wait until the process finishes (the Status will show Available).
Next, create and add an SSH key pair in Alibaba Cloud.
In the ECS Console, go to Network Security → Key Pairs:
Click Create SSH Key Pair:
Generate an SSH key pair. For details on how to generate a key pair, see . Fill in the required parameters:
Name – A custom name for your key pair.
Creation Mode – Import
Public Key – Paste your public key, generated earlier.
Resource Group – Choose your resource group.
Click OK to create the key pair in the cloud.
Before creating the virtual machine, you must set up a security group (firewall).
Go to Network & Security → Security Groups:
Click Create Security Group:
Specify the following security group parameters:
Security Group – A custom name for your security group.
Network – Your selected network. If it doesn’t exist yet, click "Create VPC" to the right.
Security Group – Basic Security Group.
Resource Group – Your resource group.
Click Create Security Group.
Go to Instances & Images → Instances:
Click Create Instance to create a new virtual machine:
Select your VM parameters:
Billing Method – Choose how you’ll pay for the VM.
Region, Network, and Zone – Select the region and zone to match your needs.
Instance – Pick a configuration for your VM.
Configure additional VM parameters:
Image – Custom Images → Choose the previously imported image.
Storage – Select the type and size of the System Disk (20 GB is the minimum for Alibaba Cloud).
Add a second disk by clicking Add Data Disk, specifying its type and size.
Choose the network parameters for your VM. The security group created earlier will be assigned automatically:
Click Create Order.
In the Instances section, open the newly created VM by clicking its name.
Click Connect.
Select VNC. A new tab will open in your browser with console access.
Enter the following command to connect via SSH:
Replace:
C:\Users\username\.ssh\id_ed25519 with the path to your SSH key,
root if you changed the default user when creating the VM,
ip-adress with the external IP address of your MikoPBX instance.
You will then connect via SSH:
On the VM’s main page, you’ll see important parameters for logging into the MikoPBX web interface.
Paste the IP address into your browser’s address bar to access the MikoPBX web interface login page.
Description and configuration of network interfaces
The "Network Interface" section in MikoPBX is an interface for configuring the system's network connection parameters. Here, administrators can manage IP addresses, subnet masks, gateways, and other network settings for each network interface. This allows MikoPBX to be correctly integrated into the organization's network and ensure its stable operation in accordance with the requirements of the network infrastructure.
The section is located in "Network and Firewall" -> "Network Interface":
The hostname is the name of the machine. If no value is specified, the default hostname used is 'mikopbx.local'.
There are two ways to configure the IP address:
DHCP (Dynamic Host Configuration Protocol) can be used for automatic IP address configuration. Enable the 'Use DHCP to obtain network settings' switch. This is recommended for most users. To not rely on DHCP server settings (to provide a specific address), you can disable the switch.
If you do not want to use settings obtained from a DHCP server, you can configure the network manually. This requires some knowledge about the network topology. To the right of the IP address, there is a field for Subnet Mask in CIDR format. You should use the alternative format: /8 corresponds to the subnet mask 255.0.0.0, /16 corresponds to 255.255.0.0, and /24 corresponds to 255.255.255.0.
"VLAN ID" - MikoPBX supports virtual network interfaces. This is relevant only for physical PCs. Sometimes a PC may have only one network interface, and it may not be possible to connect a second one physically. Using VLAN, you can create a virtual interface that works 'on top' of the physical one. One of the advantages of using VLAN is that all phone calls can be routed through it, while the network equipment can 'tag' all VLAN traffic and guarantee a stable connection.
The number of network interfaces in MikoPBX is not limited.
The 'Network interface with internet access' is the primary network interface through which access to external addresses (non-local) will be established.
If no DNS server addresses are specified, MikoPBX uses built-in fallback DNS servers. For non-Russian locales, the order is 1.1.1.1, 8.8.8.8, and then 77.88.8.8; for the Russian locale, 77.88.8.8 is tried first, followed by 1.1.1.1 and 8.8.8.8.
Depending on your network topology, you need to perform the following steps to configure MikoPBX. The PBX can be behind a network router, which is the most common scenario, or it can have a public IP.
If the PBX is behind a router, you need to check the 'This station is located behind a NAT router' option.
If you know the external address of the station (IP or domain name) and have forwarded the ports of the PBX to the external world, it is recommended to fill in the fields 'External IP address of your router' or 'External hostname of your router'.
For all addresses that are not local to the PBX, the station will be represented by the external address:
If 'External IP address of your router' is empty and 'External hostname of your router' is filled, the PBX will be represented by the hostname (External hostname) field.
Static routes are used when MikoPBX must send traffic to a specific network through a separate gateway instead of the main Internet gateway. This is most often required in the following cases:
the PBX has several network interfaces;
telephony, VPN, or remote offices are available through a separate router;
traffic must be routed to a specific provider or branch-office subnet;
the main gateway is used for Internet access, while separate internal networks must be reached through another gateway.
Go to Network and Firewall → Network interfaces.
At the bottom of the form, find the Static routes block.
In the Static routes block, click Add route.
Fill in the route row.
Add several routes if needed. You can change the order by dragging rows; this order is saved as the route priority.
Click Save at the bottom of the form.
Example route to a remote subnet:
Example route to a single address:
Check that the required remote address or service is reachable from MikoPBX: SIP provider registration, connection to a remote phone, access to a VPN network, or another working scenario for which the route was added.
If you have console or SSH access, you can additionally check which path the system selects:
The output should show the expected gateway or interface. To check host reachability, you can use:
You can also open Maintenance → System logs → System information and check the current network settings.
Invalid gateway. The Gateway field must contain a valid IP address. If it is incorrect, MikoPBX may show the Invalid gateway address message.
The gateway is not in the selected interface subnet. If a specific Interface is selected, make sure the specified gateway is reachable through that interface. Otherwise, the route may be saved, but traffic will not go through the required path.
Conflict with the main gateway. Do not use a static route as a replacement for the main Internet gateway. The main gateway is configured in the Internet interface settings, while static routes are better suited for separate networks and addresses.
Creating and Configuring Dialplan Applications
Dialplan applications are programmable voice applications in PHP and Asterisk Dialplan. MikoPBX comes with several pre-configured applications. With some basic knowledge of Asterisk dialplans, additional applications can be easily created. Like a phone account, applications can have an extension assigned in the settings.
MikoPBX comes with several pre-configured applications. With some basic knowledge of Asterisk dialplan, you can easily create additional applications. Like a phone extension, applications can have an internal number assigned in the settings.
Below you will see a description of the basic applications included in MikoPBX:
MikoPBX applications are created from several plans of the Asterisk application suite. There are many examples of ready-to-run applications in the system. To add a new MikoPBX application, click on "Add a New" in the application menu.
In just a few steps, you can create your own applications. First, define the Name and Call Number for the application, and optionally fill in the Comment field.
Possible application code types:
PHP-AGI script - AGI is an embedded method in Asterisk for executing external scripts (similar to CGI for HTTP servers), which can extend Asterisk's functionality using other programming languages, particularly PHP. AGI scripts can control call handling in the dialplan and are invoked from the extensions.conf file.
Asterisk Dialplan - The configuration of the dialplan is contained in the Asterisk configuration file called extensions.conf. This is one of the most important configuration files where the processing and routing of incoming and outgoing calls are defined. This file governs the behavior of all connections passing through your PBX (Private Branch Exchange).
Let's clarify: we will refer to MikoPBX applications as "applications" and Asterisk dialplan functions as "functions". For example, Answer(), NoOP(), Set(), and Wait() are functions. These are individual target functions in Asterisk that are then combined in MikoPBX to create more powerful MikoPBX applications.
Describe the logical operations in the text field of the Programme Code. Please note that only one command is allowed per line, for example:
The figure shows an example of the simplest application for the number 000063. After dialing the number, you will hear the robot voice your internal number.
Description of Asterisk functions that you can use in your applications:
Instructions for deploying and connecting local MinIO S3 to MikoPBX
This guide describes the process of deploying local MinIO S3 storage and connecting it to MikoPBX. All actions are performed on macOS - for other operating systems, the official documentation is available at the link.
Open the terminal and run the command:
brew install minio/stable/minioAfter the installation is complete, the terminal will display a confirmation:
🍺 /opt/homebrew/Cellar/minio/RELEASE.2025-09-06T17-38-46Z_1: 4 files, 108.2MB, built in 3 secondsCreate a directory for data.
mkdir -p ~/minio-dataStart the server:
You will see the following output with useful information, including the WebUI address for convenient further work with the storage.
After starting, leave the terminal window open - closing it will stop the server.
To start MinIO automatically when the system starts, run:
Open a browser and go to the WebUI address.
Enter the credentials:
Username: minioadmin
Password: minioadmin
Click "Create Bucket" in the left menu.
Enter a bucket name, for example mikopbx-recordings.
Click "Create Bucket".
To create an access key and configure access permissions, you will need the mc console utility.
Install MinIO Client:
After installation, check the version:
Add the local MinIO server as an alias:
Check the connection:
If the connection is successful, information about the MinIO server will appear in the terminal.
It is not recommended to use the MinIO root user for connecting external applications.
Create a separate user for MikoPBX:
Create a policy file:
Add the following content to it:
Save the file:
Create the policy in MinIO:
Attach the policy to the user:
Create an access key for the mikopbx-user user:
MinIO will display access data in response:
Save Access Key and Secret Key. They will be needed to connect MinIO to MikoPBX.
Secret Key is displayed only when it is created.
Go to the "Maintenance" -> "Storage" tab.
Open the "Cloud storage S3" tab and fill in the following fields:
Automatic recording upload to cloud storage - enable the switch.
S3 storage provider - MinIO
S3 endpoint URL - http://192.168.64.1:9000
S3 region
Configure the "Local storage period (S3 mode)" slider - choose how long recordings will be stored locally before being deleted after uploading to the cloud.
Click "Save".
After saving the settings, click "Test connection". If the connection is successful, the message "S3 connection successful" will appear and synchronization of call recordings will begin.

























































Connecting and configuring telephony providers in MikoPBX
"Telephony Providers" in MikoPBX is a section of the system where connections to external telecom operators via Internet protocols for IP telephony are configured. Here, administrators can add and configure SIP trunk accounts or other types of connections that allow the system to make and receive calls from landline and mobile numbers.
To make or receive external phone calls via the public switched telephone network or the Internet, you must create at least one provider account. Each technology has its own account type. To add a new account or change an existing one, go to "Call Routing" -> "Telephony Providers":
The provider overview contains a list of all available service providers. A green icon next to the provider's name indicates that MikoPBX has registered this provider, and you can start using this provider. You can enable or disable the use of the provider using the switch on the left.
To connect a new provider account, click Connect SIP or Connect IAX depending on the type of account you are connecting.
How to protect MikoPBX from hacking and unauthorized access
IP PBX systems are increasingly being targeted by attackers. Criminals gain access to your telephony and make calls at your expense - to premium numbers and international destinations. This can result in losses of tens or hundreds of thousands of dollars within just a few hours.
Beyond direct financial losses, a compromised PBX can be used by fraudsters to make calls on behalf of your organization - for example, calling citizens while impersonating banks or government agencies. Victims see your company's phone number, causing reputational damage and potential investigations by law enforcement.
Go through every item in this guide - even if you have already configured the system, something may have been missed.
If you are running version 2024.1.114, install the patch with a single command:
You can read more about it .
If you are using an older version — upgrade to the latest release. Steps 3 and 4 from the list above must be completed regardless of your version.
https://s3.us-west-1.wasabisys.com
eu-central-1 (Amsterdam)
https://s3.eu-central-1.wasabisys.com
eu-central-2 (Frankfurt)
https://s3.eu-central-2.wasabisys.com
eu-west-1 (London)
https://s3.eu-west-1.wasabisys.com
eu-west-2 (Paris)
https://s3.eu-west-2.wasabisys.com
ap-northeast-1 (Tokyo)
https://s3.ap-northeast-1.wasabisys.com
ap-northeast-2 (Osaka)
https://s3.ap-northeast-2.wasabisys.com
ap-southeast-1 (Singapore)
https://s3.ap-southeast-1.wasabisys.com
ap-southeast-2 (Sydney)
https://s3.ap-southeast-2.wasabisys.com













































control stream file
Sending a preset audio file to the channel, with the ability to control its playback (pause/rewind/resume playback) using the DTMF digits received from the subscriber, if specified. (Asterisk 1.2)
database del
Deleting a key/value from the database.
database deltree
Deleting the key/value tree from the database.
database get
Get the value from the database.
database put
Adding/changing a value in the database.
exec
Execution of the specified Command. (Commands are functions that you use when describing the set plan in the extensions.conf file).
get data
Get data from the channel.
get option
Behaves similarly to the "STREAM FILE" command, but is used with a specified value for timeout. (Asterisk 1.2)
get variable
Get the value of the channel variable.
hangup
Break the connection (Hangup) on the current channel.
noop
An empty command. Does nothing.
receive char
Accepts one character from the channel if it supports this feature.
receive text
Accepts a text string from a channel if it supports this feature.
record file
Writes to the specified file.
say alpha
Pronounces the specified string of characters. (Asterisk 1.2)
say date
Pronounces the date. (Asterisk 1.2)
say datetime
Pronounces the date and time according to the specified format. (Asterisk 1.2)
say digits
Pronounces the specified string of digits.
say number
Pronounces the specified number.
say phonetic
Pronounces the specified string of characters.
say time
Pronounces the time.
send image
Sends the image to the channel if it supports this feature.
send text
Sends text to the channel if it supports this feature.
set autohangup
Automatic termination of the connection (Autohangup) on the channel at the specified time.
set callerid
Setting the caller id for the current channel.
set context
Setting the context for the current channel.
set extension
Change the extension for the current channel.
set music
Включение/Выключение музыки ожидания (Music on hold), например: «SET MUSIC ON default».
set priority
Enabling/Turning off the standby music (Music on hold), for example: "SET MUSIC ON default".
set variable
Setting the channel variable.
stream file
Sending an audio file to the channel.
tdd mode
Setting the TDD mode for a channel that can support it to enable interaction with TDD.
verbose
Writing a message to the verbose log of the asterisk server.
wait for digit
Waiting for the DTMF button to be pressed
000063
The application reads the internal number of the employee used to call the application and voices it to the employee, i.e. the employee is voiced his internal number on the PBX
000064
0000MILLI - Generates a constant sound signal with a frequency of 1000 Hz. Used to check the quality of the connection.
10003246
The Echo application sends the received audio signals back to the user so that the delay duration can be determined. In general, you hear what you say. The application is mainly used for testing.
answer
Transfer the call to the answered state.
channel status
Returns the status of the connected channel.
MikoPBX will check the commands used. It is possible that incorrectly programmed operations may affect the performance of your telephone system.






Interface
Network interface through which the route will be used. You can leave Auto so that the operating system chooses the interface, or select a specific interface such as eth0 or a VLAN interface.
Description
Administrator comment. Use it to describe the purpose of the route, for example Branch VPN or SIP provider network.
Interface
eth0 or Auto
Description
Office VPN
Interface
eth0 or Auto
Invalid mask/CIDR. Specify the network address and the correct Mask. For example, for the 192.168.10.0/24 subnet, use 192.168.10.0 and 24 - 255.255.255.0; for a single IP address, use 32 - 255.255.255.255.
The route is not applied after saving. Check that the page was saved without errors, the route remains in the Static routes table, the selected gateway is reachable from MikoPBX, and Interface is selected correctly or set to Auto.
Network
Destination network address. For example, specify 192.168.10.0 for a remote subnet. If the route is needed for one specific IP address, specify that IP address and select the 32 - 255.255.255.255 mask.
Mask
CIDR mask of the destination network. It is selected from a list in the interface, for example 24 - 255.255.255.0 for a /24 subnet or 32 - 255.255.255.255 for a single address.
Gateway
Network
192.168.10.0
Mask
24 - 255.255.255.0
Gateway
Network
54.246.198.136
Mask
32 - 255.255.255.255
Gateway






IP address of the router through which MikoPBX should send traffic to the specified network. The gateway must be reachable from the selected interface.
172.16.32.15
172.16.32.15
us-east-1S3 bucket name - specify the bucket name created in MinIO (mikopbx-recordings in this guide)
Access key and Secret key - paste the values obtained in the previous part of this guide (key pair).







ip route get 192.168.10.10ping 192.168.10.10MINIO_ROOT_USER=minioadmin MINIO_ROOT_PASSWORD=minioadmin \
minio server ~/minio-data --console-address ":9001"MinIO Object Storage Server
Copyright: 2015-2026 MinIO, Inc.
License: GNU AGPLv3 - https://www.gnu.org/licenses/agpl-3.0.html
Version: RELEASE.2025-09-06T17-38-46Z (go1.24.6 darwin/arm64)
API: http://192.168.100.114:9000 http://192.168.64.1:9000 http://127.0.0.1:9000
RootUser: minioadmin
RootPass: minioadmin
WebUI: http://192.168.100.114:9001 http://192.168.64.1:9001 http://127.0.0.1:9001
RootUser: minioadmin
RootPass: minioadmin
CLI: https://docs.min.io/community/minio-object-store/reference/minio-mc.html#quickstart
$ mc alias set 'myminio' 'http://192.168.100.114:9000' 'minioadmin' 'minioadmin'
Docs: https://docs.min.iobrew services start minio/stable/miniobrew install minio/stable/mcmc --versionmc alias set local http://127.0.0.1:9000 minioadmin minioadminmc admin info localmc admin user add local mikopbx-user 'strong-password'nano mikopbx-recordings-policy.json{
"Version": "2012-10-17",
"Statement": [
{
"Effect": "Allow",
"Action": [
"s3:ListBucket",
"s3:GetBucketLocation"
],
"Resource": [
"arn:aws:s3:::mikopbx-recordings"
]
},
{
"Effect": "Allow",
"Action": [
"s3:PutObject",
"s3:GetObject",
"s3:DeleteObject"
],
"Resource": [
"arn:aws:s3:::mikopbx-recordings/*"
]
}
]
}Ctrl + O
Enter
Ctrl + Xmc admin policy create local mikopbx-recordings-rw mikopbx-recordings-policy.jsonmc admin policy attach local mikopbx-recordings-rw --user mikopbx-usermc admin accesskey create local/ mikopbx-userAccess Key: A6FMEQOVJAK7K6TUI5O4
Secret Key: HuKLsNh7hmS+3xdLzlpYaFjxu5wVxLPkM8XUTgsj
Expiration: NONE
Name:
Description:OS Type – Linux
OS Version – Others Linux
Architecture – 64-bit OS
Uncheck "Check After Import"
Allow all inbound connections (see example below). Outbound is allowed by default.
ssh -i C:\Users\username\.ssh\id_ed25519 root@ip-adressThe bucket region for your image must match the region of your future virtual machine!




























In the general settings of the SIP provider, specify the following settings:
Provider Name - an arbitrary name that is convenient for you. It will be displayed in the selection lists in the corresponding menus.
Account Type - the type of registration for the provider account.
Provider host URL or IP Address - can be either a URL or an IP address.
Username and Password provided by your provider.
DTMF Mode - determines how DTMF signals are transmitted over SIP. There are different standards used to transmit DTMF to SIP providers. Try using different standards to find the mode that suits you.
inband sends keypresses as "tones." To use this standard, you need a high-quality audio codec.
Auto, rfc, and info transmit keypresses through SIP encoding.
In this section, list all communication service provider addresses from which incoming calls can arrive. Access to these addresses for SIP and RTP ports will be automatically opened on the firewall.
By default, it is set to 5060. The SIP protocol describes how a client application (e.g., a softphone) can request the initiation of a connection from another, possibly physically remote client in the same network using its unique name. The protocol defines how clients agree on opening exchange channels based on other protocols that can be used for direct information transmission (e.g., RTP).
Allows you to specify the transport protocol used for this provider account.
This is the provider's SIP proxy server for processing requests. The actual SIP server may differ from this address. The outbound proxy takes on primary requests and forwards them to the appropriate server.
When this option is enabled, Asterisk will send SIP OPTIONS packets. This is necessary to support NAT tunneling on your router.
Specify the frequency with which Asterisk will send OPTIONS-type SIP messages to check if this device is working and available for making calls.
If this device does not respond within the specified period (default is 60 seconds), Asterisk considers it turned off and unavailable for making calls.
You can disable the use of the fromuser field of the SIP packet header.
The fromuser and fromdomain parameters in the pjsip.conf file are used for outgoing calls from Asterisk to the SIP device.
You can override:
the username in the From field in SIP packets (fromuser).
the domain name in the From field in SIP packets (fromdomain).
The fields User and Domain serve this purpose.
In this field, you can modify Asterisk configuration files.
You can override almost all parameters. For example, when using chan_pjsip, the provider is described with the following sections:
To override fields in sections, fill in the Additional Parameters field as follows:
To complete the configuration, click Save Settings.
There are cases when you need to connect multiple accounts from one communication service provider. In this case, the settings Host or IP Address and SIP Connection Port may be the same for all accounts.
Asterisk handles this situation differently. The PBX will not be able to correctly route the call to the desired provider, and the call will be dropped.
As a solution, in older versions of the PBX, you could describe additional inbound routes for which you would fill in the Additional Number (DID) field with the Username value for each account of the provider. This required creating N number of additional routes, equal to the number of provider accounts.
An alternative is the "Registering multiple accounts from one provider" instruction. This approach was not very intuitive.
The Username field, in most cases, will be used as the destination number Additional Number (DID) for incoming calls. Considering that outgoing routes for all Usernames will be configured, the call will be correctly processed by the PBX.
This option is used when connecting most providers.
Registration is necessary when the provider cannot know from which IP address the client will connect. For example, when the PBX is behind NAT. The provider's server is usually on a public IP address.
This option is relevant for the operation of some FXO / GSM gateways when an external device must connect to your PBX using a login and password.
This option is also relevant when the remote device is behind NAT, and MikoPBX cannot know its IP address.
Relevant for secure private networks. For example, Rostelecom often lays its network cable and connects the client to its local network.
In this case, the PBX and the provider must be in the same network.


[REG-AUTH-***]
type = auth
; ----
[REG-***]
type = registration
transport = transport-udp
; ----
[***-OUT]
type = auth
; ----
[***]
type = aor
max_contacts = 1
; ----
[***]
type = identify
; ----
[***]
type = endpoint
context = incoming
; ----[registration-auth]
; Describe authentication parameters for [REG-AUTH-***]
[registration]
; Describe registration parameters on the remote server [REG-***]
[endpoint-auth]
; Describe authentication parameters for outgoing calls through the provider
[identify]
; This section is responsible for matching registration and endpoint. When an incoming call arrives,
; an identity parameter check will be carried out according to the INVITE.
[aor]
; Edit the AOR section for the endpoint
[endpoint]
; Edit endpoint parametersGo to Network and Firewall → Network Firewall, make sure the toggle is enabled, and create rules that allow access only from the required subnets.
Addresses to add to your rules:
Your office subnet
VPN server addresses
Your telephony provider's IP addresses (check with your provider)
Static IP addresses of remote employees
The PBX admin panel is effectively the "master key" to the entire system. If it is accessible from the internet without restrictions, an attacker can gain full control over your telephony.
In the firewall rules, allow WEB and CTI access only for your office subnet or VPN. For all other rules, uncheck the WEB and CTI boxes. If you need remote access — use a VPN.
A weak password is the most common cause of a breach. Attackers try thousands of combinations per second, and passwords like 1234, admin, or password are cracked instantly.
Password requirements for SIP accounts and the web interface:
Minimum 12 characters
UPPER and lowercase letters
Numbers and special characters (!@#$%^&*)
No dictionary words, names, or dates of birth
What to check:
Open each employee's profile under Telephony → Extensions and verify that the SIP password is sufficiently complex.
Check the web interface password under System → General Settings → WEB interface password.
By default, the employee's extension number (e.g., 204) is used for SIP authentication. Attackers know this and specifically target standard extension numbers.
Auth Username is the username that a phone or softphone sends when registering with the PBX. It differs from the internal extension number and is used solely for authenticating the connection.
How to configure the Auth Username prefix in MikoPBX:
Go to System → General Settings → SIP and fill in the Auth Username prefix for authorization field. For example, with the prefix MIKO, extension 204 will authenticate as 204MIKO.
After changing the Auth Username, you must update the settings on every phone or softphone. The setting name varies by manufacturer:
Yealink
Register Name / Authentication User
Grandstream
Authenticate ID
Fanvil
This setting is typically found under the Account or SIP Account section in the phone's web interface.
Fail2Ban automatically blocks IP addresses that exhibit suspicious connection attempts.
Go to Network and Firewall → Intrusion Protection and review the configured protection level:
Weak — 20 attempts in 10 min, ban for 10 min. For initial setup and trusted networks.
Normal — 10 attempts in 1 hour, ban for 1 day. Recommended for most deployments.
Strong — 5 attempts in 6 hours, ban for 7 days. For internet-facing servers.
Paranoid — 3 attempts in 24 hours, ban for 30 days. For servers under active attack.
Warning: Make sure your office addresses are added to the whitelist to avoid accidentally locking yourself out.
Fail2Ban does not replace strong passwords - even with Fail2Ban enabled, a weak password can still be brute-forced.
Docker deployment: in bridge mode the built-in firewall and fail2ban do not protect the web interface. Set up an external firewall bouncer or switch the container to network_mode: host.
If your PBX is directly accessible from the internet, it becomes a target for automated scanners that continuously search for vulnerable systems.
Place the PBX behind a NAT router.
Use VPN connections for remote employees.
If a public IP is unavoidable — be sure to configure the Firewall and Fail2Ban.
Under Network and Firewall → Network interfaces, correctly specify the network topology and external address.
Even with strong technical security, it is worth adding a financial safety net. If a breach does occur, these measures will limit potential losses.
Contact your telephony provider and request:
A daily spending limit on outbound calls
A block on service when the balance is negative
Blocking of international and premium-rate calls if you do not use them
Top up your balance in small amounts as needed.
Set up spending alerts with your provider if that option is available.
If you discover that your PBX has been compromised, follow these steps:
Step 1 — Isolate the PBX Immediately
Block all external access via the firewall. Change all passwords — SIP accounts, web interface, SSH.
Step 2 — Save Logs and Call Recordings
Save call recording files and system logs separately — they may be needed as evidence. They can be overwritten over time.
Step 3 — Notify Your Telephony Provider
Contact your telephony provider and report the incident. The provider may be able to block further calls and officially document the breach.
Step 4 — Report the Incident to the Relevant Authorities
File a report with your national cybercrime authority or law enforcement agency. Briefly describe what happened, state that calls were made without your knowledge, and indicate that you are prepared to provide logs and call recordings as evidence.
Go through this list and confirm that every item has been completed:
Firewall — configuring access rules.
Network Interfaces — network configuration, NAT, DNS.
Static Routes — route configuration through the web interface.
— managing accounts and SIP passwords.
curl -L 'https://files.miko.ru/s/DPZcM2vywc2BTOZ/download' | sh⚠️ Critical vulnerability in version 2024.1.114!
A vulnerability has been discovered in the external panel module: if the module is exposed to the internet or the Firewall is misconfigured, an attacker can obtain all SIP credentials and make calls on behalf of your company.
You must perform the following steps:
Upgrade to version 2026.1.223 or newer.
Install the security patch (see below).
Close WEB, CTI, and SIP access to the PBX from external networks.
Update all passwords.
Note: For remote employees with dynamic IPs, we recommend purchasing a static IP address from their ISP (typically very low cost per month). An alternative is VPN: all remote employees connect through a VPN server, and only that server's address is added to the Firewall.

























ℹ️ Available starting from MikoPBX 2026.1.76.
Technical reference for developers writing their own bouncers and for integrating MikoPBX with edge providers (Cloudflare, AWS WAF, custom nftables generators).
Base URL: https://<MIKOPBX-HOST>/pbxcore/api/v3/firewall-bouncer/
Endpoints:
GET /v1/decisions/stream — full decision snapshot in CrowdSec LAPI format. Stock cs-firewall-bouncer polls this path automatically by appending it to api_url.
GET /v1/whitelist — operator allow-list as a flat JSON array (MikoPBX extension, see ).
Auth: the bouncer token can be sent in either header:
X-Api-Key: <token> — what stock cs-firewall-bouncer sends.
Authorization: Bearer <token> — convenient for curl, Postman, or custom HTTP clients. Both forms validate the same
Permission scope: firewall_bouncer. Issue tokens with allowed_paths: {"/api/v3/firewall-bouncer": "read"} — the UI preset in the ApiKeys section does this for you.
Compatibility: the decisions/stream response shape matches the CrowdSec LAPI exactly ({new, deleted} at the top level, no envelope). Existing CrowdSec bouncers (cs-firewall-bouncer, cs-cloudflare-bouncer, cs-nginx-bouncer, and dozens of community plugins) work out of the box.
GET /pbxcore/api/v3/firewall-bouncer/v1/decisions/stream
Every poll returns the full snapshot of currently active bans in new. Bouncers refresh their ipset / nftables entry timeouts on every appearance, so a still-active ban stays alive at the source's declared duration.
deleted carries the per-bouncer diff since the previous poll. MikoPBX stores the last-served snapshot per ApiKey id (Redis key _PH_REDIS_CLIENT:fwbouncer:cursor:<token-id>, TTL 1 h refreshed on every poll — the cursor expires only after one hour of bouncer silence). Decisions that disappeared between two polls — operator-triggered unban, ban TTL elapsed, NetworkFilters entry deleted — appear in deleted
The bouncer-sent query parameters behave as follows:
Cursor isolation lets multiple independent bouncers (e.g. one running nftables locally, another driving a Cloudflare endpoint) each track their own delta — issue one ApiKey per bouncer. Sharing one ApiKey between two or more bouncers makes them share a single cursor: polls from different bouncers interleave their snapshot writes, producing nondeterministic deleted[] timing and missed eviction events for short-lived bans.
Note that the JSON body is the LAPI shape at the top level — there is no {result, data, ...} envelope around new / deleted.
GET /pbxcore/api/v3/firewall-bouncer/v1/whitelist returns the operator-defined allow-list as a flat JSON array:
Sources merged into the response:
NetworkFilters rows with newer_block_ip = '1',
Fail2BanRules.whitelist.
This endpoint is MikoPBX-specific. Stock cs-firewall-bouncer does not poll it — CrowdSec LAPI has no "allow" decision type and the bouncer uses its own whitelists.yaml. Provided for MikoPBX-aware integrations that want server-side whitelist consistency.
MikoPBX additionally enforces the whitelist on the write side (via DockerNetworkFilterService::isIpWhitelisted), but that's defence in depth — your bouncer should still subtract the whitelist from new before applying bans.
The whitelist endpoint exposes operator allow-list networks. Never expose this endpoint without authentication.
Optimally, bind the bouncer token to a NetworkFilter that permits only the bouncer host's IP. Calls from any other source will return 403.
After compromise — revoke the token in the ApiKeys UI; the bouncer will stop receiving new decisions, and its local bans will start expiring after 30 seconds (if that mode is enabled).





























ℹ️ Available starting from MikoPBX 2026.2.76. On earlier versions the
firewall-bouncerLAPI endpoint, the Firewall page banner, and the "Create bouncer token" button do not exist.
In Docker mode, MikoPBX's built-in firewall and fail2ban do not protect the web interface:
The container cannot manage host iptables.
With Docker's default userland-proxy=true
Full installation guide for MikoPBX using AWS
Sign in to the service Amazon Web Services
To follow the instructions, install the Amazon Command Line Utility by opening Terminal and entering the following command
Let's get started with the setup
Go to your account
From the dropdown menu, select Security credentials
If you don't have an access key, do the following












Authentication User
Snom
Authentication Username
Linphone
Auth userid
Zoiper
Authentication user / Auth. Username
MicroSIP
Login
Cisco (SPA)
Auth ID

ApiKeystype
string
Always "ban".
scope
string
"Ip" for a single address or "Range" for a CIDR.
value
string
IP address or CIDR.
duration
string
Remaining time until expiry, e.g. "3600s" or "8760h".
scenario
string
mikopbx/sip, mikopbx/http, mikopbx/ami, mikopbx/iax, or mikopbx/manual.
mikopbx/sip
scenario tag
Redis sip category
mikopbx/http
scenario tag
Redis http category
mikopbx/ami
scenario tag
Redis ami category
mikopbx/iax
scenario tag
Redis iax category
mikopbx/manual
scenario tag
Operator-defined Firewall UI entries
update_frequency: 5–10s is the recommended polling interval.
id
int
Stable positive 32-bit integer — crc32(value + scenario) & 0x7fffffff.
origin
string
mikopbx-fail2ban
aggregated origin
Redis keys firewall:<cat>:<ip>
mikopbx-networkfilters
aggregated origin
startup
startup=true ignores the stored cursor for this poll only — full snapshot in new, empty deleted — then writes the just-served snapshot to the cursor. The NEXT poll diffs against that fresh snapshot normally. Bouncers send this on the first poll after restart. Other values (including false) are treated as steady-state polls.
scopes
Accepted but ignored. MikoPBX only emits scope=Ip and scope=Range; no filtering applied server-side.
origins
mikopbx-fail2ban (Redis) or mikopbx-networkfilters (operator DB).
m_NetworkFilters.deny table
Accepted but ignored. Both origins (mikopbx-fail2ban, mikopbx-networkfilters) always present in the response. Bouncers that want to filter must do so client-side.
{
"new": [
{
"id": 12345,
"origin": "mikopbx-fail2ban",
"type": "ban",
"scope": "Ip",
"value": "203.0.113.7",
"duration": "3600s",
"scenario": "mikopbx/sip"
},
{
"id": 678901,
"origin": "mikopbx-networkfilters",
"type": "ban",
"scope": "Range",
"value": "198.51.100.0/24",
"duration": "8760h",
"scenario": "mikopbx/manual"
}
],
"deleted": []
}# Read the bouncer API key from an env var / secret manager on the host
# where you run this — never paste it inline in scripts that get checked
# into version control.
TOKEN=$BOUNCER_API_KEY
# CrowdSec-style:
curl -H "X-Api-Key: $TOKEN" \
"https://pbx.example.com/pbxcore/api/v3/firewall-bouncer/v1/decisions/stream?startup=true" \
| jq
# Bearer-style (equivalent):
curl -H "Authorization: Bearer $TOKEN" \
"https://pbx.example.com/pbxcore/api/v3/firewall-bouncer/v1/decisions/stream" \
| jqimport requests, time
BASE = "https://pbx.example.com/pbxcore/api/v3/firewall-bouncer"
HEADERS = {"X-Api-Key": TOKEN}
# Bouncer-local whitelist (refreshed less often than decisions).
whitelist = set(requests.get(f"{BASE}/v1/whitelist", headers=HEADERS).json())
while True:
resp = requests.get(f"{BASE}/v1/decisions/stream", headers=HEADERS).json()
bans = {d["value"] for d in resp["new"]}
apply_iptables(bans - whitelist)
time.sleep(10)["10.0.0.0/8", "192.168.1.0/24"]docker0172.17.0.1SIP protection still works: UDP DNAT preserves the source IP, Asterisk sees the real address, fail2ban writes the ban to Redis, and module reload acl rejects subsequent REGISTERs. Only the HTTP segment is broken.
The fix is to export ban decisions outside the container and apply them in the real host firewall (or edge CDN, or cloud security group) via an external bouncer.
The Security → Web access page shows a yellow banner — "Docker bridge: external firewall enforcement required" — when MikoPBX detects the failure mode. If you see it, this document is for you.
The Check my IP visibility button calls the system:checkClientIpVisibility endpoint and reports one of three verdicts:
ip_visible — the real client IP is visible; no action needed.
ip_not_visible — the real client IP has been replaced by the Docker bridge gateway. HTTP firewall rules will not protect you.
proxy_detected — a reverse proxy is in front of the PBX, and the PBX deliberately does not trust proxy headers. Configure the proxy to expose the real source IP, or deploy an external bouncer.
If the host is dedicated to the PBX and there are no port conflicts, flip the container to host mode:
The container shares the host network namespace; Asterisk and Nginx see real source IPs, and the built-in firewall works as on bare metal. Best for SIP-heavy installations.
Limitations: only one host-mode container per host, no side-by-side PBX copies, conflicts with other processes on standard ports.
The MikoPBX container stays in bridge mode. On the Linux host, install cs-firewall-bouncer (open-source, CrowdSec project). It polls the MikoPBX endpoint every 10 seconds and translates decisions into the host's iptables / nftables.
Recommended for most installations.
Open System → API keys.
Click Create bouncer token (pre-fills the correct path restriction).
A token-creation form opens. The description ("External firewall bouncer (CrowdSec-compatible)") and the API key are pre-filled — optionally pick a Network filter to restrict the source IP the bouncer is allowed to call from. Leave Full access permissions off: the Create bouncer token button has already scoped the token to /api/v3/firewall-bouncer, so it has no access to the rest of the API.
Save. A modal will pop up with a ready-to-paste cs-firewall-bouncer.yaml snippet — copy it immediately, the API key is shown only once.
Open /etc/crowdsec/bouncers/cs-firewall-bouncer.yaml and replace api_url / api_key with the values from step 1:
📌
api_urlis the base URL — cs-firewall-bouncer appends/v1/decisions/streamitself and sends the token in theX-Api-Keyheader. Do not put the full decisions path inapi_url, and do not prefix the key withBearer— the bouncer manages both.
⚠️ If your MikoPBX listens on HTTPS with a self-signed certificate, add
insecure_skip_verify: trueor install the CA certificate on the host.
🚨
iptables_chains: [INPUT, FORWARD, DOCKER-USER]is not the CrowdSec default (default isINPUTonly). WithoutDOCKER-USER, traffic Docker routes to the container goes via theDOCKERchain and never sees the bouncer's DROP rule — the ban appears in iptables but actually does nothing. This is the single most common trap when wiring CrowdSec to a Docker-hosted PBX.
The bouncer log should show received N new decisions, 0 deleted.
sudo iptables -L CROWDSEC -n (or the IPv6 counterpart for crowdsec-firewall-bouncer-iptables-v6) lists the applied bans.
Manually ban a test IP via the Firewall → Networks UI or trigger a fail2ban ban, and confirm the entry appears in the host iptables within 30 seconds.
CrowdSec bouncers ban at IP level, not protocol level — a single ipset entry drops every TCP and UDP packet from the banned address. That includes port 22. If an operator's source IP ends up in the ban list by mistake (say, fail2ban detects three failed auth:login attempts from the office NAT), SSH gets dropped too, and the operator can lock themselves out of the host.
Insert a high-priority ACCEPT for the admin port above the bouncer's DROP rule, so administrative access stays reachable even when the operator's own IP gets banned:
Repeat for any other port you administer through (Wireguard, the cloud provider's serial console, etc.) — anything you do not want the bouncer to ever drop.
For installations where losing access to the host has a high cost, add a systemd-timer that periodically flushes the bouncer's ipset. The bouncer will re-apply current bans on the next poll, so this is a bounded-blast-radius safety net rather than a feature disable:
30 minutes is a reasonable default — short enough that an accidental lockout self-recovers before the operator's coffee gets cold, long enough that real attacks still get blocked for a meaningful window.
CrowdSec maps decisions to a single ipset per IP family. iptables rules then drop all traffic from listed IPs, regardless of protocol or port. So a mikopbx/http ban (web brute-force) will also silently drop the same IP's SIP / IAX / AMI / SSH packets — and the reverse: a mikopbx/sip ban (Asterisk fail2ban jail) will drop the same IP's HTTP and AMI too. MikoPBX feeds all four ban categories (sip, http, ami, iax) into the same stream, so once the bouncer is wired every ban becomes IP-wide.
For most installations that's the right behaviour — if an IP is hostile to HTTP it has no business reaching SIP either, and vice versa. But if you deliberately want per-protocol isolation (e.g. block HTTP brute-forcers without affecting their SIP), do not deploy the bouncer for that PBX. The existing in-Docker SIP defence path (fail2ban → Redis → pjsip ACL inside Asterisk) keeps SIP bans isolated to Asterisk, but only as long as those bans never reach the host ipset — which is exactly what the bouncer changes.
GET /pbxcore/api/v3/firewall-bouncer/v1/decisions/stream returns a snapshot of currently active decisions in the exact shape stock cs-firewall-bouncer expects — {new, deleted} at the top level, no MikoPBX envelope:
new[] carries the full snapshot of currently active bans on every poll — bouncers refresh their entry timeouts to the value of duration, so an active ban stays alive at the source's declared lifetime. deleted[] is computed per-token (MikoPBX stores the previous snapshot per ApiKey id) and contains decisions that disappeared since the last poll. Operator-triggered unbans propagate to the bouncer's ipset in one poll cycle (≈ 5–10 seconds), not at natural-TTL decay.
?startup=true on the first poll after bouncer restart resets the per-token cursor and emits deleted: [] — so a freshly-restarted bouncer never sees phantom evictions for state it never tracked.
Both header forms authenticate the same token:
GET /pbxcore/api/v3/firewall-bouncer/v1/whitelist returns the operator-defined whitelist as a flat JSON array:
This endpoint is MikoPBX-specific. Stock cs-firewall-bouncer does not poll it (CrowdSec LAPI has no "allow" decision type, and the bouncer uses its own whitelists.yaml). Provided for MikoPBX-aware integrations that want server-side whitelist consistency with the PBX's NetworkFilters.
The full response format, query parameters, and MikoPBX↔CrowdSec field mapping are documented in the firewall-export endpoint reference.
services:
mikopbx:
image: mikopbx/mikopbx:latest
network_mode: host
# delete all `ports:` entries# Debian / Ubuntu
curl -s https://install.crowdsec.net | sudo sh
sudo apt-get install -y crowdsec-firewall-bouncer-iptablesapi_url: http://<MIKOPBX-HOST>/pbxcore/api/v3/firewall-bouncer/
api_key: <token-from-modal>
update_frequency: 10s
mode: iptables
log_mode: stdout
log_level: info
# CRITICAL for Docker deployments: the rule MUST be inserted in
# DOCKER-USER, otherwise traffic Docker forwards to the container
# bypasses INPUT/FORWARD entirely and the ban is silently
# ineffective.
iptables_chains:
- INPUT
- FORWARD
- DOCKER-USER
# Disable IPv6 unless Docker has an IPv6 bridge configured. ip6tables
# does not have a DOCKER-USER chain on hosts without docker IPv6, and
# the bouncer fatals at startup when it tries to insert there.
disable_ipv6: truesudo systemctl restart crowdsec-firewall-bouncer.service
sudo systemctl status crowdsec-firewall-bouncer.service# Always-accept SSH (port 22) — survives bouncer state changes
sudo iptables -I INPUT 1 -p tcp --dport 22 -j ACCEPT \
-m comment --comment "admin-protect"
# Persist across reboots (Debian / Ubuntu)
sudo apt-get install -y iptables-persistent
sudo netfilter-persistent save# /etc/systemd/system/crowdsec-safety-flush.service
[Unit]
Description=Safety net - flush crowdsec ipset to prevent lockout
After=crowdsec-firewall-bouncer.service
[Service]
Type=oneshot
ExecStart=/usr/sbin/ipset flush crowdsec-blacklists# /etc/systemd/system/crowdsec-safety-flush.timer
[Unit]
Description=Run crowdsec ipset flush every 30 minutes
[Timer]
OnBootSec=30min
OnUnitActiveSec=30min
Unit=crowdsec-safety-flush.service
[Install]
WantedBy=timers.targetsudo systemctl daemon-reload
sudo systemctl enable --now crowdsec-safety-flush.timer{
"new": [
{
"id": 12345,
"origin": "mikopbx-fail2ban",
"type": "ban",
"scope": "Ip",
"value": "203.0.113.7",
"duration": "3600s",
"scenario": "mikopbx/sip"
}
],
"deleted": []
}# Stock cs-firewall-bouncer (CrowdSec convention):
curl -H "X-Api-Key: <token>" \
"http://<MIKOPBX-HOST>/pbxcore/api/v3/firewall-bouncer/v1/decisions/stream?startup=true"
# Equivalent for ad-hoc probes with curl / Postman / Insomnia:
curl -H "Authorization: Bearer <token>" \
"http://<MIKOPBX-HOST>/pbxcore/api/v3/firewall-bouncer/v1/decisions/stream"["10.0.0.0/8", "192.168.1.0/24"]Under the Access keys table, select Create access key
Copy the Access key and Secret access key
If you already have an access key, simply copy the Access key and Secret access key
Open Services / Storage / S3
On the tab select Create bucket
Enter a unique bucket name
Use default values for other fields
After entering the values, click Create bucket
Open the created bucket and select Upload
On the opened tab select Add files
Upload the file from the MikoPBX distribution with the .raw extension
Click Upload
Wait for the file to finish uploading
If not done previously for this cloud
Create a separate folder for files on your computer
Create a file named trust-policy.json in the folder
Open Terminal and navigate to the created folder
Run the command vi trust-policy.json
Enter editing mode by pressing i and paste the text
Press ESC and type :wq to save the file
Similarly, create a file named role-policy.json and change the bucket name value in the text to the name of your created bucket
Similarly, create a file named import-image.sh, change the DEFAULT_BUCKET parameter value to the name of your created bucket and the DEFAULT_IMAGE parameter value to the name of the image uploaded to the bucket
Run the command aws configure, specify the region and copied Access key and Secret access key
Run the command
Run the command
Run the command
If the command executes successfully, a unique AMI identifier will be generated
Open Services / Compute / EC2 and navigate to Images / AMIs
Select the created image and click Launch an instance from AMI to create a virtual machine
Enter the virtual machine name, for example mikopbx-vm
Specify the instance type - t3.micro
If you have an SSH key
Specify the SSH key in the Key pair field
If you don't have an SSH key
Select Create new key pair and specify the key pair name, for example mikopbx_key
Follow the instructions further
In the Network settings section, check Allow SSH traffic and Allow HTTPS traffic
If necessary, change the size of the storage disk in Configure storage, default size is 50Gb
For other fields use default values
Click Launch instance
Go to the created virtual machine mikopbx-vm
On the opened tab, select Connect / EC2 serial console, wait for the system to fully load until the authentication parameters are displayed
Copy the external address of the created virtual machine and enter it in the browser's address bar
Use the login and password provided in EC2 serial console for login
curl "https://awscli.amazonaws.com/AWSCLIV2.pkg" -o "AWSCLIV2.pkg"
sudo installer -pkg AWSCLIV2.pkg -target /vi trust-policy.json{
"Version": "2012-10-17",
"Statement":
[
{
"Effect": "Allow",
"Principal": { "Service": "vmie.amazonaws.com" },
"Action": "sts:AssumeRole",
"Condition": {
"StringEquals":{"sts:Externalid": "vmimport"}
}
}
]
}{
"Version":"2012-10-17",
"Statement":
[
{
"Effect": "Allow",
"Action": [
"s3:GetBucketLocation",
"s3:GetObject",
"s3:ListBucket"
],
"Resource": [
"arn:aws:s3:::mikopbx-bucket",
"arn:aws:s3:::mikopbx-bucket/*"
]
},
{
"Effect": "Allow",
"Action": [
"ec2:ModifySnapshotAttribute",
"ec2:CopySnapshot",
"ec2:RegisterImage",
"ec2:Describe*"
],
"Resource": "*"
}
]
}
#!/bin/bash
# Default variable definition
DEFAULT_IMAGE="mikopbx-2024.1.40-dev-x86_64.raw"
DEFAULT_BUCKET="mikopbx-bucket"
DEFAULT_DESCRIPTION="MikoPBX the best open source PBX on asterisk"
DEFAULT_NAME="MikoPBX 2024.1.40-dev"
# Overriding variables with environment variable values, if set
IMAGE="${IMAGE:-$DEFAULT_IMAGE}"
BUCKET="${BUCKET:-$DEFAULT_BUCKET}"
DESCRIPTION="${DESCRIPTION:-$DEFAULT_DESCRIPTION}"
NAME="${NAME:-$DEFAULT_NAME}"
# JSON file for import-snapshot command
JSON_FILE="disk_container.json"
# Creating JSON file
cat <<EOF> ${JSON_FILE}
{
"Description": "${DESCRIPTION} image",
"Format": "raw",
"UserBucket": {
"S3Bucket": "${BUCKET}",
"S3Key": "${IMAGE}"
}
}
EOF
# Importing the snapshot
IMPORT_TASK_ID=$(aws ec2 import-snapshot --description "${DESCRIPTION} image" --disk-container "file://${JSON_FILE}" --query 'ImportTaskId' --output text)
echo "Import task started with ID: $IMPORT_TASK_ID"
# Waiting for snapshot import to complete
while true; do
STATUS=$(aws ec2 describe-import-snapshot-tasks --import-task-ids $IMPORT_TASK_ID --query 'ImportSnapshotTasks[0].SnapshotTaskDetail.Status' --output text)
echo "Current status: $STATUS"
if [ "$STATUS" == "completed" ]; then
break
fi
sleep 30
done
# Getting SnapshotId
SNAPSHOT_ID=$(aws ec2 describe-import-snapshot-tasks --import-task-ids $IMPORT_TASK_ID --query 'ImportSnapshotTasks[0].SnapshotTaskDetail.SnapshotId' --output text)
# Registering AMI
AMI_ID=$(aws ec2 register-image \
--name "$NAME" \
--description "$DESCRIPTION" \
--architecture x86_64 \
--sriov-net-support simple \
--virtualization-type hvm \
--ena-support \
--boot-mode legacy-bios \
--root-device-name "/dev/sda1" \
--block-device-mappings "[{\"DeviceName\": \"/dev/sda1\", \"Ebs\":{\"DeleteOnTermination\":true, \"VolumeSize\":1, \"SnapshotId\":\"$SNAPSHOT_ID\"}}, {\"DeviceName\": \"/dev/sdb\", \"Ebs\":{\"VolumeSize\":50}}]" \
--query 'ImageId' \
--output text)
echo "AMI created with ID: $AMI_ID"
aws configureaws iam create-role --role-name vmimport --assume-role-policy-document "file://trust-policy.json"aws iam put-role-policy --role-name vmimport --policy-name vmimport --policy-document "file://role-policy.json"sh import-image.shTo deploy the PBX use two disks:
A 1 Gb disk for the main system
A 50+ Gb disk for storing call recordings
Make sure to configure the Firewall on the MikoPBX
Setting Primary Phone Numbers
Extensions in MikoPBX are individual users of the system who are assigned internal numbers for making and receiving calls. They have personal accounts that allow you to configure access rights, call forwarding and other personal settings in the system.
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, click the "Add new employee" button.
There is an option to export and import employees for configuration convenience. To do this, click the arrow to the right of the "Add New Employee" button.
3 options are available:
Import from CSV — load employees from a CSV file into MikoPBX.
Export to CSV — download employees to a CSV file from MikoPBX (employees will not be deleted from the station).
Download template — download a CSV table template to fill in and subsequently import into MikoPBX.
Click "Select CSV file" and choose the previously prepared file with data in the table. It is recommended to use templates from the "Template" tab.
After selecting the file, information about all detected users in the table will be displayed. Select a duplicate handling strategy and click "Confirm import" to start the process.
After the process is complete, you will see the employee creation status as well as a notification about the end of the import.
Click "Back to list" to return to the employee list.
There is an option to export a CSV file with all the data of current employees. Several export formats are available:
Minimal:
number — Internal number (required)
user_username — Employee full name (required)
user_email — Email address
mobile_number — Mobile number
Standard:
All parameters from Minimal.
mobile_dialstring — Number for dialing mobile
sip_dtmfmode — DTMF mode (auto/rfc4733/info/inband/auto_info)
sip_transport — Transport (udp/tcp/tls), default udp,tcp
Full:
All parameters from Minimal and Standard.
user_avatar — Photo URL
sip_acceptMultipleCalls — Accept multiple calls at the same time (true/false)
sip_manualattributes — Additional SIP parameters
You can also specify a range of internal employee numbers to export (the "Filter by number range" section).
Click "Export employees". The file will be downloaded to your device.
On this tab, you can download a blank file template with the specified "columns" to fill in and subsequently import into MikoPBX.
Select the template format (see the "" section for more details), then click "Download CSV template".
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.
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.
Yealink T19
Yealink T21
Yealink T28
Snom D120
MikoPBX Installation deployment Guide using Google Cloud
Authorize on the platform
Let's start configuring
Open Navigation menu / Products & solutions / Storage / Cloud Storage
On the Cloud Storage tab select Create
Enter the storage name, for example
Updating MikoPBX in a Proxmox VE LXC container without copying call recordings
This guide is intended for MikoPBX installed in a Proxmox VE LXC container. All operations are performed through the Proxmox and MikoPBX web interfaces.
In LXC, the MikoPBX system files and user data are stored on separate volumes:
MikoPBX Installation Guide using Docker compose
To work with MikoPBX in a container, you need to install Docker and Docker Compose following the instructions
Here is an example of a docker-compose.yml file that can be used to manage your MikoPBX container via Docker Compose:
Save the contents into a file named docker-compose.yml, make the necessary adjustments, and launch MikoPBX using the command:
It is also possible to organize the launch of multiple MikoPBX containers on a single host. However, you need to consider Docker's port handling features. If the –net=host mode is not used, it will lead to a high load on the host system's CPU because Docker creates a separate rule in Iptables for each allocated port.
With the –net=host mode enabled, you need to manually monitor the distribution of available ports between the running containers and built-in applications. For instance, to run two MikoPBX containers on one host, you can use the following configuration file:
Save the contents into a file named docker-compose.yml
Description of the main system settings
This section configures the core system parameters. It is recommended to complete these settings immediately after installing the PBX.
PBX system name - displayed on the MikoPBX main page.
Additional description - visible to system administrators only.

















sip_secret — SIP password (will be generated if not specified)
fwd_ringlength — Ring time (seconds) before forwarding
fwd_forwarding — Forwarding number if no answer
sip_enableRecording — Call recording (true/false)
fwd_forwardingonbusy — Forwarding number if busy
fwd_forwardingonunavailable — Forwarding number if unavailable
Password for SIP
Configuring telegram as SIP softphone
Please set strong passwords for SIP accounts. MikoPBX validates the password length and strength when the account is saved. A secure SIP password should use a non-dictionary combination of uppercase and lowercase letters, digits, and special characters.
By setting complex passwords, you can enhance the security of the user accounts and protect them from unauthorized access.



















[***]
type = aor
max_contacts = 10
; ----
[***]
type = auth
; ----
[***]
type = endpoint
context = all_peers
; ----
[acl_***]
deny = 0.0.0.0/0.0.0.0
permit = 0.0.0.0/0.0.0.0
; ----[acl]
; Describe access parameters from different subnets [acl_***]
[auth]
; Describe authentication parameters for outbound calls
[aor]
; Edit AOR section for the endpoint
[endpoint]
; Edit endpoint parametersUse default values for other fields
After entering the values, click the Create button
Open the created miko-images storage
On the opened tab select UPLOAD FILES
Upload the file from the MikoPBX distribution with the .vhd extension
Wait for the file upload to complete
Open Navigation menu / APIs & Services / Library / Google Enterprise API / Cloud Build API
On the opened tab select ENABLE
Open Navigation menu / APIs & Services / Library / Google Enterprise API / Compute Engine API
On the opened tab select ENABLE
Open Navigation menu / Products & solutions / Compute / Compute Engine
Go to the Storage / Images section
Select CREATE IMAGE to create a new image
Enter the image name, for example mikopbx-new-image
Specify the source type - Virtual disk (VMDK, VHD)
Select Virtual disk file under the BROWSE link, Browse / miko-images / .vhd
Uncheck the Install guest packages checkbox
In the Operating system on virtual disk field specify - No operating system. Data only.
Use default values for other fields
After entering the values, click the Create button and wait for the image creation to complete
In the Compute Engine tab go to the Virtual machines / VM Instance section
Select CREATE INSTANCE
Enter the virtual machine name, for example mikopbx-vm
In the Machine configuration / General purpose table select Series - N1
In the Machine type section choose Shared-core / f1-micro from the dropdown menu
In the Boot disk section select CHANGE
On the opened tab go to CUSTOM IMAGES
In the Image field select the previously created mikopbx-new-image
Use default values for other fields on the tab
Click the SELECT button
In the Advanced options / Disks section select ADD NEW DISK
On the opened tab enter the disk name for data storage, for example disk-storage-mikopbx
Specify the disk size of at least 50GB
Use default values for other fields on the tab
Click the SAVE button
If you have an SSH key, proceed as follows:
In the Advanced options / Security / MANAGE ACCESS section, select ADD ITEM
Specify it in the SSH key field
If you don't have an SSH key, proceed directly to step 18
Use default values for other fields
After entering the values, click the CREATE button
Open Navigation menu / VPC network / Firewall
Select CREATE FIREWALL RULE to create a new rule for incoming connections
Enter the name of the new rule, for example internal-allow
Specify Direction of traffic - Ingress
Specify Targets
From the Source filter dropdown menu select IPV4 ranges and in the Source IPv4 ranges field enter 0.0.0.0/0
In the Protocols and ports section check the TCP (Ports - 0-65535) and UDP (Ports - 0-65535) checkboxes
Use default values for other fields
After entering the values, click the CREATE button
Open the Compute Engine tab and go to the Virtual machines / VM Instance section
Go to the created virtual machine mikopbx-vm
On the opened tab navigate to Logs / Serial port 1 (console)
Copy the external address of the created virtual machine and enter it in the browser's address bar
For login use the login and password provided in Serial port 1 (console)
Авторизуйтесь на платформе https://console.cloud.google.com/
Приступим к настройке
Откройте Navigation menu / Products & solutions / Storage / Cloud Storage
На вкладке Cloud Storage выберите Create
Введите имя хранилища, например miko-images
Для других полей используйте значения по умолчанию
Завершив ввод значений, нажмите кнопку Create
Откройте созданное хранилище miko-images
На открывшейся вкладке выберите UPLOAD FILES
Загрузите файл из дистрибутива MikoPBX с расширением .vhd
Дождитесь окончания загрузки файла
Откройте Navigation menu / APIs & Services / Library / Google Enterprise API / Cloud Build API
На открывшейся вкладке выберите ENABLE
Откройте Navigation menu / APIs & Services / Library / Google Enterprise API / Compute Engine API
На открывшейся вкладке выберите ENABLE
Откройте Navigation menu / Products & solutions / Compute / Compute Engine
Перейдите в раздел Storage / Images
Выберите CREATE IMAGE для создания нового образа
Введите имя образа (Name), например mikopbx-new-image
Укажите тип источника (Source) - Virtual disk (VMDK, VHD)
Выберите Virtual disk file по ссылке BROWSE, *Browse / miko-images / .vhd
Уберите флажок Install guest packages
В поле Operating system on virtual disk укажите - No operating system. Data only.
Для других полей используйте значения по умолчанию
Завершив ввод значений, нажмите кнопку Create и дождитесь окончания создания образа
Во вкладке Compute Engine перейдите в раздел Virtual machines / VM Instance
Выберите CREATE INSTANCE
Введите имя виртуальной машины (Name), например mikopbx-vm
В таблице Machine configuration / General purpose ****выберите ****Series - N1
В разделе Machine type в выпадающем меню выберите Shared-core / f1-micro
В разделе Boot disk выберите CHANGE
На открытой вкладке перейдите к CUSTOM IMAGES
В поле Image выберите созданный ранее образ mikopbx-new-image
Для других полей на вкладке используйте значения по умолчанию
Нажмите кнопку SELECT
В разделе Firewall ****разрешите HTTP трафик, установите флажок Allow HTTP traffic
В разделе Advanced options / Disks выберите ADD NEW DISK
На открытой вкладке введите имя диска для хранения данных (Name), например disk-storage-mikopbx
Укажите размер диска (Size) не менее 50Гб
Для других полей на вкладке используйте значения по умолчанию
Нажмите кнопку SAVE
Если у вас есть ключ SSH, выполните следующее
В разделе Advanced options / Security / MANAGE ACCESS выберите ADD ITEM
Укажите его в поле SSH key
Если у вас есть нет ключа SSH, сразу перейдите к пункту 19
Для других полей используйте значения по умолчанию
Завершив ввод значений, нажмите кнопку CREATE
Откройте Navigation menu / VPC network / Firewall
Выберите CREATE FIREWALL RULE для создания нового правила для входящего соединения
Введите имя нового правила (Name), например http-80
Укажите Direction of traffic - Ingress
Укажите Targets
Из выпадающего меню в поле Source filter выберите IPV4 ranges и в поле Source IPv4 ranges укажите 0.0.0.0/0
В разделе Protocols and ports установите флажок TCP и в поле Ports укажите 80
Для других полей используйте значения по умолчанию
Завершив ввод значений, нажмите кнопку CREATE
Аналогично создайте правило для HTTPS. Укажите имя, Direction of traffic - Ingress, Targets, Source filter - IPV4 ranges, Source IPv4 ranges - 0.0.0.0/0, Protocols and ports - TCP, Ports - 443
Аналогично создайте правило для SIP сигнализации TCP. Укажите имя, Direction of traffic - Ingress, Targets, Source filter - IPV4 ranges, Source IPv4 ranges - 0.0.0.0/0, Protocols and ports - TCP, Ports - 5060
Аналогично создайте правило для SIP сигнализации UDP. Укажите имя, Direction of traffic - Ingress, Targets, Source filter - IPV4 ranges, Source IPv4 ranges - 0.0.0.0/0, Protocols and ports - UDP, Ports - 5060
Аналогично создайте правило для передачи звука RTP. Укажите имя, Direction of traffic - Ingress, Targets, Source filter - IPV4 ranges, Source IPv4 ranges - 0.0.0.0/0, Protocols and ports - UDP, Ports - 10000-10200
Откройте вкладку Compute Engine перейдите в раздел Virtual machines / VM Instance
Скопируйте External IP созданной виртуальной машины
Введите в строке браузера External IP
Откройте созданную виртуальную машину
Скопируйте Instance Id - это пароль для входа в АТС по умолчанию
Логин для входа по умолчанию - admin
To deploy the PBX use two disks:
A 1 Gb disk for the main system
A 50+ Gb disk for storing call recordings
Make sure to configure the Firewall on the MikoPBX
Remains attached to the old CT
/storage
Call recordings, call history, modules, audio files, and logs
Reassigned to the new CT
/cf
MikoPBX settings and the main database
Reassigned to the new CT
To update, create a new container from the current lxc.tar.gz template. The existing /storage and /cf volumes are not copied: Proxmox reassigns them to the new container using Reassign Owner.
As a result, the new MikoPBX version gets a new root file system while continuing to use the existing settings, history, and call recordings.
Schedule a maintenance window. After the old CT is stopped, telephony will remain unavailable until the new container starts.
Make sure you have:
administrator access to Proxmox VE;
administrator access to MikoPBX;
a backup stored outside the container being updated.
Open the old CT in Proxmox and record its settings.
Resources — CPU, RAM, Swap, and Root Disk size.
Network — bridge, IPv4/IPv6, VLAN, MAC address, and Proxmox Firewall status.
DNS — DNS servers.
Options — whether the container is privileged or unprivileged.
Under Resources, find the entries containing:
mp=/storage — the volume with recordings and user data;
mp=/cf — the volume with the configuration.
Record their identifiers. They are usually mp0 and mp1, but the order may differ.
Open Modules → Module marketplace.
Make sure the Backup module is installed and enabled.
Open the module and click Create backup copy.
Create the archive and download it to the administrator's computer or external storage.
In Proxmox, open the old container (CT) → Backup and click Backup now.
Select external or separate backup storage. Start the backup and wait for the TASK OK message.
Open the MikoPBX releases page.
Find the required version and copy the link to the file whose name ends with lxc.tar.gz.
In Proxmox, select the local storage → CT Templates and click Download from URL.
Paste the link into the URL field and click Query URL. Verify the file name and click Download.
Wait for the TASK OK message.
Click Create CT in the upper-right corner of Proxmox.
Select the same Proxmox node that hosts the old CT:
Specify a new CT ID. Do not use the ID of the old container.
Enter a different name, such as mikopbx-update-test.
Set Unprivileged container to the same value as on the old CT.
Click Next.
Select the downloaded lxc.tar.gz template for the new version and click Next.
Create a new Root Disk (rootfs). Select suitable storage and set the system disk size to 1 GB.
Click Next.
Specify the same number of CPU cores and the same Memory and Swap values as on the old container.
Reproduce the network settings of the old CT. For testing, we recommend using an available temporary IP address and a new MAC address to avoid a conflict with the old container.
If DHCP is used, the new CT may obtain a different address automatically. If the DHCP server reserves the address by MAC address, create a separate reservation for the new MAC address in advance.
Reproduce the DNS settings of the old CT:
Carefully review the settings on the Confirm page.
Clear the Start after created checkbox. The new container must not be started before /cf and /storage are attached. Click Finish and wait for the TASK OK message.
Open the new CT → Resources. At this stage, the list must contain the new Root Disk, but it must not contain /cf or /storage mount points.
Open the old CT → Resources. Select the Mount Point entry whose settings contain mp=/storage. Click Volume Action → Reassign Owner.
In the Target field, select the new CT. Leave Mount Point selected in the Add as field. Proxmox suggests an available mount point number, such as mp0. Make sure the selected number is not already in use.
Click Reassign Volume.
After the operation, the volume disappears from the old CT's Resources section and appears under the new CT. Proxmox changes the volume owner and, if necessary, renames it for the new CT ID. The contents of the disk are neither copied nor formatted.
Repeat the same procedure for the Mount Point whose settings contain mp=/cf:
Open the old CT → Resources. Select the Mount Point entry whose settings contain mp=/cf. Click Volume Action → Reassign Owner.
Select the new CT. Leave Add as → Mount Point selected. Choose the next available number, such as mp1, and click Reassign Volume.
Open the new CT → Resources and check the final configuration:
Root Disk belongs to the new CT;
one Mount Point contains mp=/storage;
the other Mount Point contains mp=/cf;
the /storage and /cf sizes match the sizes of the old volumes.
The mp0 and mp1 numbers may be in a different order. The mp=/storage and mp=/cf paths are what matter.
Select the new CT and click Start. Open the Console tab.
Wait for MikoPBX to finish booting and display the web interface address.
Open this address in a browser and sign in with the previous MikoPBX login and password.
If the new CT received the previous production IP address immediately, skip this step.
If a temporary address was used for testing:
Stop the new CT.
Make sure once again that the old CT is stopped.
Open the new CT → Network and select the net0 interface.
Enter the previous MikoPBX IPv4/IPv6 address.
If the address is reserved on the DHCP server, move the reservation to the new MAC address or assign the old MAC address to the new CT.
Check the bridge, VLAN, gateway, and Proxmox Firewall settings.
If necessary, switch external NAT, DNS, and firewall rules to the new address.
Start the new CT.
The same IP or MAC address may be used only when the old container is completely stopped.
Do not delete the old CT until all checks have been completed.
Verify in the web interface that the new MikoPBX version is displayed.
Check the extensions, SIP accounts, providers, and inbound and outbound routes.
Open Telephony → Call history and make sure the previous call history is available.
Play several old call recordings from different dates.
Check the installed modules, their licenses, and settings.
Make sure the phones and providers are registered or have the expected status.
Make an internal, outbound, and inbound test call.
Check that the new call appears in the history and that its recording can be played.
Reboot the new CT through Proxmox.
After the reboot, check the web interface, network, registrations, and a test call again.
If the new version does not work correctly:
Stop the new CT.
Under its Resources, select the /storage Mount Point.
Select Volume Action → Reassign Owner and return the volume to the old CT.
Return the /cf volume to the old CT in the same way.
Verify that mp=/storage and mp=/cf are present on the old CT again.
Restore the old IP, MAC, DNS, and NAT settings.
Start the old CT and verify its operation.
Keep the old CT stopped and retain the backups for several days. Delete the old container only after:
the new version has been running reliably;
inbound and outbound calls have been tested;
new call recordings are created and can be played;
MikoPBX has rebooted successfully;
a new backup of the updated PBX has been created.
Before deleting the old CT, open its Resources once more and make sure that /cf and /storage have actually been reassigned to the new container. The old Root Disk can then be deleted together with the old CT.
Root Disk
We recommend updating sequentially without skipping releases or versions. If a release description specifies an intermediate version, update to that version first.
System files for the current MikoPBX version
Do not update an LXC container with an .img file under Maintenance → PBX update, and do not attach an ISO image. These methods are intended for a full system disk in a virtual or physical machine. For LXC, use a new lxc.tar.gz template.
Reassign Owner does not create a backup. It only changes the owner of an existing volume. An administrator error, storage failure, or incompatible database change may require restoring from a backup.
The new version may have changed the database structure on /cf. Returning the volumes therefore does not guarantee compatibility with the old MikoPBX version. If the old CT does not start correctly, restore it from the Proxmox backup or restore the settings from the MikoPBX archive created before the update.
With this update method, years of call recordings remain on the original /storage disk. The switchover time does not depend on the volume of recordings because Proxmox changes the volume owner instead of copying its contents.
There is an option to launch MikoPBX containers in the –net=bridge mode. However, as mentioned above, to use this mode you either need to significantly limit the range of RTP ports or open them on the host machine without using Docker's capabilities.
For this, you will need to write a small script to determine the name of the current bridge interface and the IP address of each container. After running Docker Compose, you will then need to add the necessary iptables rules for the RTP port range as follows:
Let's describe several containers in the docker-compose.yaml file, specify different ports for the web interface, SIP ports, and ranges of RTP ports to ensure they do not overlap.
Creating a directory for scripts
Save the start-multiple-mikopbx.sh and docker-compose.yaml files into this folder.
Install the necessary dependencies for the script.
Navigate to our folder, add execution rights and launch our script.
While waiting for the containers to start, check the firewall settings on the host, and if necessary, open the ports specified in our docker-compose.yaml file, specifically:
TCP/UDP ports 5060 and 6060 for SIP
UDP ranges 10000-10800 and 20000-20800 for RTP voice transmission
TCP ports 8443 and 9443 for HTTPS protocol, for web interface operation.
Access each station in turn at the addresses:
https://<host machine IP>:8443
https://<host machine IP>:9443
To access the web interface of the first MikoPBX, use the login admin and the password mikopbx-first-password
To access the web interface of the second MikoPBX, use the login admin and the password mikopbx-second-password
Each machine should have NAT mode enabled, indicating that the container is behind a router in the network interface settings. If the stations will be used within a local network, then in the external IP field, enter the local IP address of the host machine, otherwise its public IP address.
With that, the setup is complete, and you can configure accounts and make calls.
Below are some of the environment variables that will allow you to adjust the MikoPBX ports and settings used.
SSH_PORT - port for SSH (22)
WEB_PORT - port for the web interface via HTTP protocol (80)
WEB_HTTPS_PORT - port for the web interface via HTTPS protocol (443)
SIP_PORT - port for connecting a SIP client (5060)
TLS_PORT - port for connecting a SIP client with encryption (5061)
RTP_PORT_FROM - beginning of the RTP port range, voice transmission (10000)
RTP_PORT_TO - end of the RTP port range, voice transmission (10800)
IAX_PORT - port for connecting IAX clients (4569)
AMI_PORT - AMI port (5038)
AJAM_PORT - AJAM port used for connecting the telephony panel for 1C (8088)
AJAM_PORT_TLS - AJAM port used for connecting the telephony panel for 1C (8089)
BEANSTALK_PORT - port for the Beanstalkd queue server (4229)
REDIS_PORT - port for the Redis server (6379)
GNATS_PORT - port for the gnatsd server (4223)
ID_WWW_USER - identifier for www-user (can be set with the expression
$(id -u www-user), where www-user is NOT a root user)
ID_WWW_GROUP - group identifier for www-user (can be set with the expression
$(id -g www-user), where www-user is NOT a root group)
WEB_ADMIN_LOGIN - login for Web interface access
WEB_ADMIN_PASSWORD - password for Web interface access
A full list of all possible setting parameters is available in the source code here.
services:
mikopbx:
container_name: "mikopbx"
image: "ghcr.io/mikopbx/mikopbx:latest"
network_mode: "host"
entrypoint: "/sbin/docker-entrypoint"
hostname: "mikopbx-in-a-docker"
volumes:
- /var/spool/mikopbx/cf:/cf
- /var/spool/mikopbx/storage:/storage
tty: true
environment:
- ID_WWW_USER=${ID_WWW_USER}
- ID_WWW_GROUP=${ID_WWW_GROUP}
# Change the station name through environment variables
- PBX_NAME=MikoPBX-in-Docker
# Change the default SSH port to 23
- SSH_PORT=23
# Change the default WEB port to 8080
- WEB_PORT=8080
# Change the default WEB HTTPS port to 8443
- WEB_HTTPS_PORT=8443export ID_WWW_USER=$(id -u www-user)
export ID_WWW_GROUP=$(id -g www-user)
sudo docker compose -f docker-compose.yml up -dservices:
mikopbx-first:
container_name: "mikopbx-first"
image: "ghcr.io/mikopbx/mikopbx:latest"
network_mode: "host"
entrypoint: "/sbin/docker-entrypoint"
hostname: "mikopbx-in-docker-first"
volumes:
- /var/spool/mikopbx/first/cf:/cf
- /var/spool/mikopbx/first/storage:/storage
tty: true
environment:
- ID_WWW_USER=${ID_WWW_USER}
- ID_WWW_GROUP=${ID_WWW_GROUP}
- PBX_NAME=MikoPBXFirst
- PBX_FIREWALL_ENABLED=0
- PBX_FAIL2BAN_ENABLED=0
- SSH_PORT=123
- WEB_PORT=8080
- WEB_HTTPS_PORT=8443
- SIP_PORT=5060
- TLS_PORT=5061
- RTP_PORT_FROM=10000
- RTP_PORT_TO=10800
- IAX_PORT=4569
- AMI_PORT=5038
- AJAM_PORT=8088
- AJAM_PORT_TLS=8089
- BEANSTALK_PORT=4229
- REDIS_PORT=6379
- GNATS_PORT=4223
mikopbx-second:
container_name: "mikopbx-second"
image: "ghcr.io/mikopbx/mikopbx:latest"
network_mode: "host"
tty: true
entrypoint: "/sbin/docker-entrypoint"
hostname: "mikopbx-in-docker-second"
volumes:
- /var/spool/mikopbx/second/cf:/cf
- /var/spool/mikopbx/second/storage:/storage
environment:
- ID_WWW_USER=${ID_WWW_USER}
- ID_WWW_GROUP=${ID_WWW_GROUP}
- PBX_NAME=MikoPBXSecond
- PBX_FIREWALL_ENABLED=0
- PBX_FAIL2BAN_ENABLED=0
- SSH_PORT=2223
- WEB_PORT=8081
- WEB_HTTPS_PORT=9443
- SIP_PORT=6060
- TLS_PORT=6061
- RTP_PORT_FROM=20000
- RTP_PORT_TO=20800
- IAX_PORT=5569
- AMI_PORT=6038
- AJAM_PORT=9088
- AJAM_PORT_TLS=9089
- BEANSTALK_PORT=5229
- REDIS_PORT=7379
- GNATS_PORT=5223export ID_WWW_USER=$(id -u www-user)
export ID_WWW_GROUP=$(id -g www-user)
sudo docker compose -f docker-compose.yml up -d#!/bin/bash
COMPOSE_FILE="$1"
if [ -z "$COMPOSE_FILE" ]; then
echo "Usage: $0 path/to/docker-compose.yaml"
exit 1
fi
# We will obtain the user ID for running the container
export ID_WWW_USER=$(id -u www-user)
export ID_WWW_GROUP=$(id -g www-user)
# Stop current containers if they are running
docker compose -f "$COMPOSE_FILE" down
# Remove them
docker compose -f "$COMPOSE_FILE" rm
# Start containers in the background
docker compose -f "$COMPOSE_FILE" up -d
sleep 60
# Create a label for IPTABLES rules
IPTABLES_COMMENT="mikopbx-custom-rule"
# Determine the project identifier, used when creating a network bridge
project_prefix=$(cat "$COMPOSE_FILE" | yq e '.x-project-name' -)
# If the prefix is not set, use a default value
if [ -z "$project_prefix" ]; then
project_prefix="default_prefix"
fi
# Function to get container IP address
function get_container_ip() {
docker inspect -f '{{range .NetworkSettings.Networks}}{{.IPAddress}}{{end}}' "$1"
}
# Function to get the name of the bridge interface
function get_bridge_name() {
local network_name="$1"
local prefix="$2"
local network_id=$(docker network inspect "${prefix}_${network_name}" -f '{{.Id}}')
if [ -z "$network_id" ]; then
echo "Error: Network ${prefix}_${network_name} not found."
return 1
fi
local bridge_name=$(ip link show type bridge | grep -o "br-${network_id:0:12}\b")
echo $bridge_name
}
echo "Delete tagged iptables rules"
# Delete all iptables rules tagged with our comment
iptables -S | grep "$IPTABLES_COMMENT" | sed 's/-A /-D /' | while read rule; do
echo "Delete rule $rule"
iptables $rule
done
# Delete all NAT iptables rules tagged with our comment
iptables -S -t nat | grep "$IPTABLES_COMMENT" | sed 's/-A /-D /' | while read rule; do
echo "Delete rule $rule"
iptables -t nat $rule
done
# Parse the docker-compose file and obtain all necessary parameters.
echo "Parsing docker-compose file and configuring iptables rules"
cat "$COMPOSE_FILE" | yq e '.services[] | select(.environment[] | test("RTP_PORT_FROM")) | {"container_name": .container_name, "environment": .environment, "network": .networks[0]}' -o=json | jq -c '.' | while read -r service; do
container_name=$(echo $service | jq -r '.container_name')
network_name=$(echo $service | jq -r '.network')
bridge_name=$(get_bridge_name "$network_name" "$project_prefix")
container_ip=$(get_container_ip "$container_name")
RTP_PORT_FROM=$(echo $service | jq -r '.environment[] | select(contains("RTP_PORT_FROM")) | split("=")[1]')
RTP_PORT_TO=$(echo $service | jq -r '.environment[] | select(contains("RTP_PORT_TO")) | split("=")[1]')
echo "Configuring iptables for $container_name ($container_ip) on $bridge_name from port $RTP_PORT_FROM to $RTP_PORT_TO"
iptables -A DOCKER -t nat ! -i "$bridge_name" -p udp -m udp --dport $RTP_PORT_FROM:$RTP_PORT_TO -j DNAT --to-destination $container_ip:$RTP_PORT_FROM-$RTP_PORT_TO -m comment --comment "$IPTABLES_COMMENT"
iptables -A DOCKER -d $container_ip/32 ! -i "$bridge_name" -o "$bridge_name" -p udp -m udp --dport $RTP_PORT_FROM:$RTP_PORT_TO -j ACCEPT -m comment --comment "$IPTABLES_COMMENT"
iptables -A POSTROUTING -t nat -s $container_ip/32 -d $container_ip/32 -p udp -m udp --dport $RTP_PORT_FROM:$RTP_PORT_TO -j MASQUERADE -m comment --comment "$IPTABLES_COMMENT"
echo "Don't forget to open UDP ports $RTP_PORT_FROM to $RTP_PORT_TO on external firewall if it exists"
done
echo "iptables configuration completed successfully."
services:
mikopbx-first:
container_name: "mikopbx-first"
image: "ghcr.io/mikopbx/mikopbx:latest"
entrypoint: "/sbin/docker-entrypoint"
hostname: "mikopbx-in-docker-first"
volumes:
- /var/spool/mikopbx/first/cf:/cf
- /var/spool/mikopbx/first/storage:/storage
tty: true
networks:
- network-bridge1
environment:
- ID_WWW_USER=${ID_WWW_USER}
- ID_WWW_GROUP=${ID_WWW_GROUP}
- PBX_NAME=MikoPBXFirst
- RTP_PORT_FROM=10000 # UDP range 10000-10800 on host will be directed to the container
- RTP_PORT_TO=10800
- WEB_ADMIN_PASSWORD=mikopbx-first-password
- ENABLE_USE_NAT=1
- PBX_FIREWALL_ENABLED=1
- PBX_FAIL2BAN_ENABLED=1
ports:
- "8443:443" # TCP port 8443 on the host is directed to port 443 in the container
- "5060:5060/udp" # UDP port 5060 on the host is directed to port 5060 in the container
mikopbx-second:
container_name: "mikopbx-second"
image: "ghcr.io/mikopbx/mikopbx:latest"
tty: true
networks:
- network-bridge2
entrypoint: "/sbin/docker-entrypoint"
hostname: "mikopbx-in-docker-second"
volumes:
- /var/spool/mikopbx/second/cf:/cf
- /var/spool/mikopbx/second/storage:/storage
environment:
- ID_WWW_USER=${ID_WWW_USER}
- ID_WWW_GROUP=${ID_WWW_GROUP}
- PBX_NAME=MikoPBXSecond
- RTP_PORT_FROM=20000 # UDP range 20000-20800 on host will be directed to the container
- RTP_PORT_TO=20800
- EXTERNAL_SIP_PORT=6060 # Inform MikoPBX about its external SIP port
- WEB_ADMIN_PASSWORD=mikopbx-second-password
- ENABLE_USE_NAT=1
- PBX_FIREWALL_ENABLED=1
- PBX_FAIL2BAN_ENABLED=1
ports:
- "9443:443" # TCP port 9443 on the host is directed to port 443 in the container
- "6060:5060/udp" # UDP port 6060 on the host is directed to port 5060 in the container
x-project-name: mikopbx # This parameter must be present
networks:
network-bridge1:
driver: bridge
network-bridge2:
driver: bridgemkdir -p /usr/src/mikopbxsudo apt-get update
sudo apt-get install jq
sudo snap install yqcd /usr/src/mikopbx
sudo chmod +x start-multiple-mikopbx.sh
sudo ./start-multiple-mikopbx.sh docker-compose.yamlImportant note! One of our containers uses port forwarding from SIP port changing its value from 5060 to 6060. In this case, for the system to function correctly, you need to add the external value of the SIP port in the NAT settings in the network interfaces section of MikoPBX. This setting can also be made by setting the corresponding value of the environment variable EXTERNAL_SIP_PORT=6060 in the docker-compose file.
In the default bridge mode the built-in MikoPBX firewall does not protect the web interface — the container cannot manage host iptables. See .
Maximum length of internal numbers - the maximum length of an employee's internal extension number.
Allow incoming calls from any servers - allows accepting SIP calls from unauthorized devices and servers without registration.
Restart PBX every night — automatic restart of Asterisk at night (at 01:00 AM system time).
Send crash information to developers — when an error occurs, its description is sent to developers (requires internet access).
Click "Save".
Call Recording - enable or disable recording of all calls.
Recording internal conversations- enable or disable recording of calls between employees.
Below, you can select audio files to be used as a recording notification (different audio files can be selected for incoming and outgoing calls).
Phone calls are saved in WebM format with the Opus codec. File size depends on call quality: if at least one participant uses a high-quality codec (e.g., G.722 or Opus), the recording is saved at a higher bitrate — this takes more disk space but improves speech recognition quality.
Call Parking is a way to temporarily place a customer on hold while you look up information. The caller hears music while waiting.
MikoPBX supports two parking methods:
Dial *2 during a call — the call will be placed on hold and you will be told the parking slot number. Any employee can pick up the call by dialing that number.
In the settings, configure a parking number — when a call is transferred to this number, MikoPBX will place it on hold and announce the slot number. Any employee can retrieve the call.
The parking slot range and parking number can be configured in this section:
Call parking number — the number to transfer a call to in order to place it on hold, default is 800.
Parking slot range — the range of parking slot numbers, default is 801–820.
MikoPBX supports two types of transfers:
Attended (Consultative) Transfer — you can speak with a colleague before transferring the call to them. The caller is on hold during this time. The transfer completes when you hang up.
Blind (Unattended) Transfer — the call is transferred immediately, without a prior conversation with the colleague. Useful when a second call comes in while you are already busy — the call can be instantly transferred to a free employee.
The key combinations for transfers can be changed in this section:
Combination for attended transfer — default is ##.
Combination for blind transfer — default is **.
Call return time if no answer after attended transfer — if no one answers after an attended transfer, the call returns. Set in seconds, default is 45 sec.
Maximum timeout between digits when entering an extension number (in milliseconds) — the wait time for the next digit when dialing an extension. Set in milliseconds, default is 2500 ms.
If a colleague's phone is ringing, you can pick up the call without leaving your desk:
*8<ColleagueNumber> — pick up a specific employee's call.
*8 — pick up any incoming call when the colleague's number is unknown.
The pickup combination can be changed in the "Combination for intercepting incoming calls" field, default is *8.
Session Initiation Protocol (SIP) is the signaling protocol used by most VoIP phones. You can change the SIP port (default 5060) to improve security.
RTP (Real-time Transport Protocol) defines the standard format for transmitting audio and video over IP networks. The default port range is 10000–10800. Some routers and firewalls may require additional range configuration. Another reason to expand the range is a large number of concurrent calls: each active call uses two RTP ports, meaning 200 ports support no more than 100 simultaneous calls. If load is higher — expand the range.
SIP port for registering phones on this station — the port for phone registration on the station, default 5060. Changing the port can improve system security.
SIP TLS port (encrypted calls) — the port for encrypted calls, default 5061.
RTP port range — the port range for audio transmission, default 10000–10800.
STUN server address — helps when the PBX is behind NAT, including when using WebRTC.
Auth Username prefix for authorization — by default, the username for SIP account authorization matches the employee's internal extension (e.g., 101). When this setting is filled in, the specified prefix will be appended to the auth username: username remains 101, but AuthUsername becomes 101MIKO. This approach significantly complicates password brute-forcing for SIP accounts.
Use WebRTC — additional settings will be applied for WebRTC connections. For example, for internal extension 201, an additional endpoint will be created, accessible via WebRTC using the URL sip:201-WS@IP_PBX.
Some firewalls close ports after a period of inactivity — in such cases, it is advisable to reduce the registration timeout. Different SIP providers may also require different timeout values.
Default time in seconds to send Keep-alive — the interval for sending keep-alive packets in seconds, default 120 seconds.
Minimum Registration Time (SIPMiniExpiry) — default 60 seconds.
Maximum Registration Time (SIPMaxExpiry) — default 3600 seconds.
This section configures the allowed audio and video codecs for the entire PBX.
Asterisk Manager Interface (AMI) is a powerful and convenient software interface (API) for Asterisk that allows external programs to manage the system. Through AMI, external programs can connect to Asterisk via TCP, initiate command execution, read results, and receive real-time event notifications. AMI is often used for integration with business processes and CRM (Customer Relationship Management) systems.
Asynchronous Javascript Asterisk Manager (AJAM) is a technology that allows web browsers or other HTTP-capable applications to directly access the Asterisk Manager Interface (AMI) via HTTP/HTTPS.
Asterisk REST Interface (ARI) is a RESTful API with WebSocket support that provides full control over Asterisk channels, bridges, and media streams in real time. Designed for developing custom telephony applications.
Use AMI Interface — enable or disable AMI.
AMI Port — the port for connecting external programs to AMI, default 5038. A client application connects to AMI through this port and authenticates, after which Asterisk responds to requests and sends notifications about state changes in specified subsystems.
HTTP Port (AJAM and ARI) — the port for HTTP connections, default 8088.
HTTPS Port (AJAM and ARI) — the port for HTTPS connections, default 8089.
Use AJAM Interface — enable or disable AJAM.
Use ARI Interface — enable or disable ARI. Disabled by default.
CORS allowed origins — domains from which requests to ARI are permitted. CORS is a browser security mechanism that restricts cross-domain API requests.
SSH (Secure Shell) is an encrypted protocol commonly used for interacting with and remotely managing servers. An SSH server can authenticate users using various algorithms. The most popular is password authentication. It is fairly simple but not very secure: passwords are transmitted over a secure channel, but are not complex enough to withstand brute-force attempts. The computational power of modern systems combined with specialized scripts makes brute-forcing very easy.
A more secure authentication method is SSH keys. Each pair consists of a public and private key: the private key is stored on the client, and the public key is uploaded to the server in the ~/.ssh/authorized_keys file. When connecting, the server sends a message encrypted with the public key — if the client decrypts it with the private key and returns the correct response, authentication is considered successful.
SSH port — the port for SSH connections, default 22.
SSH console login — the username for connecting.
Disable password authentication — enabled by default in MikoPBX (password authentication is disabled).
SSH password — the login password (available only if password authentication is not disabled).
Authorized SSH Keys — add your public SSH key here using the "+ Add Key" button. If you have multiple keys, add each one separately.
System Public SSH Key — the public SSH key of the current PBX. It can be copied into the "Authorized SSH Keys" field on another station — this allows connecting to the remote server without additional authentication.
To improve security, you can change the HTTP port (default 80) or enable HTTPS mode. HTTPS encrypts traffic between the browser and the PBX using SSL/TLS protocols. The default TCP port is 443.
HTTP port — the port for accessing the web interface via HTTP, default 80.
HTTPS port — the port for accessing the web interface via HTTPS, default 443.
Redirect to HTTPS — when the web interface is opened via HTTP, the user will be automatically redirected to HTTPS.
An SSL/TLS certificate is a digital document that verifies the server's identity and ensures encrypted communication between the browser and the PBX. In MikoPBX, the certificate is used for:
HTTPS access to the web interface
WebRTC connections (required for browser-based calls)
Secure AJAM and ARI connections via HTTPS
Secured REST API for integrations
The certificate must be in PEM format — beginning with -----BEGIN CERTIFICATE----- and ending with -----END CERTIFICATE-----. If you have intermediate certificates, add them after the main certificate in the same field.
Ways to obtain a certificate:
Let's Encrypt Module — automatic issuance and renewal of free certificates. Recommended method.
Purchase from a Certificate Authority (DigiCert, Comodo, GlobalSign, etc.)
Self-signed Certificate — automatically generated on first PBX startup, but causes browser warnings.
The secret key used to decrypt SSL/TLS connections. It must exactly match the public certificate — if they do not match, HTTPS will not work.
The key must be in PEM format — beginning with -----BEGIN RSA PRIVATE KEY----- or -----BEGIN PRIVATE KEY-----.
Security recommendations:
Keep a backup copy of the key in a secure location.
Use keys of at least 2048 bits in length (4096 recommended).
Regularly renew certificates and keys.
In this section, you can change the login and password for accessing the web interface, and configure login via Passkeys.
Login — the username for logging into the web interface.
Password — the password for logging into the web interface.
Passkeys are a modern passwordless login method using biometrics or a hardware security key: Face ID, Touch ID, Windows Hello, or YubiKey. This is faster and more secure than traditional passwords.
To add a Passkey, click the "+ Add Passkey" button and follow the browser instructions.
This section allows you to fully reset the system to its factory state. The reset will permanently delete all settings, call history, call recording files, and installed extension modules.
To confirm, type delete everything in the input field and click "Save".

Never use * in production. Only specify trusted domains over HTTPS.
Short description of ARI (Asterisk REST Interface)
ARI is a RESTful API with WebSocket support that gives full control over Asterisk channels, bridges, and media streams in real time. Unlike the MikoPBX REST API, ARI works directly with the Asterisk core and is designed for building custom telephony applications.
ARI is used when the standard PBX features are not enough and custom call handling logic is required:
WebRTC applications and softphones — web phones and mobile clients with direct media stream control
Interactive voice response (IVR) — custom menu logic unavailable through the standard dialplan
Conference calls — programmatic control of bridges and participants
Call recording and processing — real-time audio interception for analytics or transcription
Voice bots and assistants — integration with external AI services
Advanced queues — custom call distribution logic
ARI consists of three components:
REST API — management of Asterisk objects: channels, bridges, recordings
WebSocket (/asterisk/ari/events) — receiving asynchronous real-time events: incoming call, call end, DTMF keypress, etc.
Stasis — a dialplan application that passes a channel to your ARI application for control
Typical scenario: a call enters the dialplan → Stasis() passes the channel to your application → the application controls the call via REST API and receives events via WebSocket.
Before starting, you need to enable the ARI interface (it is disabled by default). Go to "System" → "General Settings".
Go to the "AMI & ARI" tab and toggle the "Use ARI Interface" switch. In the "CORS allowed origins" field, specify the domains from which requests to ARI will be made. CORS is a browser security mechanism that restricts cross-domain API requests.
Go to "System" → "ARI Access".
Click "Add User".
Fill in the following parameters:
Username — login for connection, e.g. ari_user.
Password — password for connection.
Description — description for the current user, e.g. "WebRTC Demo".
Save the settings.
Replace [application] with the name of your Stasis application.
Authentication: HTTP Basic Auth — ARI user login and password.
This is a minimal ARI example — a channel enters a Stasis application, the application plays a sound file and ends the call.
The example is taken from the official Asterisk documentation:
In MikoPBX, go to "Routing" → "Dialplan Applications", create an application with the type "Asterisk Dialplan" and the following code:
Assign the application to the required incoming route.
When an incoming call arrives, the WebSocket will receive a StasisStart event:
Open a new terminal window and run the following command:
On successful playback, you will see the following output in the terminal:
After the call ends, the WebSocket will send a StasisEnd event:
A live employee status table in the terminal — no incoming route or Stasis application configuration required. Works by subscribing to all station events.
Install dependencies:
As calls are made, the table will update in real time:
Full ARI documentation is available on the Asterisk website:

































*
All origins — for testing purposes only
(empty)
Access from the same domain only
http://localhost:3000
Local development
https://app.mycompany.com
Regular
ws://your-mikopbx.com:8088/asterisk/ari/events
Secure (TLS)
wss://your-mikopbx.com:8089/asterisk/ari/events
HTTP
http://your-mikopbx.com:8088/asterisk/ari
HTTPS
https://your-mikopbx.com:8089/asterisk/ari/
Never use * in production. Only specify trusted domains over HTTPS.





Production application
# REST API request via curl
curl -u username:password https://your-mikopbx.com:8089/asterisk/ari/asterisk/infowscat -c "wss://username:password@your-mikopbx.com:8089/asterisk/ari/events?app=hello-world"1,Answer()
n,Stasis(hello-world)
n,Hangup(){
"type": "StasisStart",
"timestamp": "2026-03-25T07:18:27.423+0300",
"args": [],
"channel": {
"id": "mikopbx-1774412307.28",
"name": "Local/10003258@internal-incoming-0000000a;2",
"state": "Up",
"protocol_id": "",
"caller": {
"name": "79257184275",
"number": "79257184275"
},
"connected": {
"name": "",
"number": "252"
},
"accountcode": "",
"dialplan": {
"context": "internal",
"exten": "10003258",
"priority": 2,
"app_name": "Stasis",
"app_data": "hello-world"
},
"creationtime": "2026-03-25T07:18:27.371+0300",
"language": "en-en"
},
"asterisk_id": "82:6a:9e:68:10:11",
"application": "hello-world"
}curl -u username:password -X POST \
"https://your-mikopbx.com:8089/asterisk/ari/channels/mikopbx-1774412307.28/play?media=sound:/storage/usbdisk1/mikopbx/media/custom/miko_hello"{
"id": "a0ee0d43-2af5-4250-a303-43825507a06c",
"media_uri": "sound:/storage/usbdisk1/mikopbx/media/custom/miko_hello",
"target_uri": "channel:mikopbx-1774412307.28",
"language": "en-en",
"state": "playing"
}{
"type": "StasisEnd",
"timestamp": "2026-03-25T07:19:44.982+0300",
"channel": {
"id": "mikopbx-1774412307.28",
"name": "Local/10003258@internal-incoming-0000000a;2",
"state": "Up",
"protocol_id": "",
"caller": {
"name": "79257184275",
"number": "79257184275"
},
"connected": {
"name": "",
"number": "252"
},
"accountcode": "",
"dialplan": {
"context": "internal",
"exten": "10003258",
"priority": 2,
"app_name": "Stasis",
"app_data": "hello-world"
},
"creationtime": "2026-03-25T07:18:27.371+0300",
"language": "en-en"
},
"asterisk_id": "82:6a:9e:68:10:11",
"application": "hello-world"
}pip install requests websocketsimport asyncio
import websockets
import json
import os
from datetime import datetime
ARI_HOST = 'your-mikopbx.com'
ARI_USER = 'ari_user'
ARI_PASS = 'your-ari-password'
peers = {}
STATES = {
'NOT_INUSE': ('🟢', 'Available'),
'BUSY': ('🔴', 'Busy'),
'UNAVAILABLE': ('⚫', 'Unavailable'),
}
def draw():
print('\033[2J\033[H', end='')
now = datetime.now().strftime('%H:%M:%S')
print(f'MikoPBX — Presence Monitor [{now}]')
print('─' * 50)
print(f' {"Number":<10} {"Name":<20} {"Status":<15} {"Updated"}')
print('─' * 50)
for number, info in sorted(peers.items()):
icon, label = STATES.get(info['state'], ('❓', info['state']))
print(f' {number:<10} {info["name"]:<20} {icon} {label:<12} {info["updated"]}')
print('─' * 50)
print(f' Employees: {len(peers)}')
async def run():
uri = (
f"wss://{ARI_USER}:{ARI_PASS}@{ARI_HOST}:8089/asterisk/ari/events"
f"?app=auto-receptionist&subscribeAll=true"
)
async with websockets.connect(uri) as ws:
draw()
async for message in ws:
event = json.loads(message)
etype = event.get('type')
if etype == 'DeviceStateChanged':
ds = event.get('device_state', {})
name = ds.get('name', '')
state = ds.get('state', '')
if not name.startswith('PJSIP/'):
continue
number = name.replace('PJSIP/', '')
if number not in peers:
peers[number] = {'name': number, 'state': state, 'updated': '—'}
peers[number]['state'] = state
peers[number]['updated'] = datetime.now().strftime('%H:%M:%S')
draw()
elif etype == 'PeerStatusChange':
ep = event.get('endpoint', {})
number = ep.get('resource', '')
state = ep.get('state', '')
if not number:
continue
if number not in peers:
peers[number] = {'name': number, 'state': 'unknown', 'updated': '—'}
if state == 'online':
peers[number]['state'] = 'NOT_INUSE'
elif state == 'offline':
peers[number]['state'] = 'UNAVAILABLE'
peers[number]['updated'] = datetime.now().strftime('%H:%M:%S')
draw()
elif etype == 'ContactStatusChange':
ep = event.get('endpoint', {})
number = ep.get('resource', '')
ci = event.get('contact_info', {})
status = ci.get('contact_status', '')
if not number:
continue
if number not in peers:
peers[number] = {'name': number, 'state': 'unknown', 'updated': '—'}
if status == 'Reachable':
peers[number]['state'] = 'NOT_INUSE'
elif status in ('Unreachable', 'NonQualified'):
peers[number]['state'] = 'UNAVAILABLE'
peers[number]['updated'] = datetime.now().strftime('%H:%M:%S')
draw()
asyncio.run(run())MikoPBX — Presence Monitor [11:34:43]
──────────────────────────────────────────────────
Number Name Status Updated
──────────────────────────────────────────────────
202 202 🟢 Available 11:34:25
243 243 ⚫ Unavailable 11:34:43
252 252 🔴 Busy 11:34:41
──────────────────────────────────────────────────
Employees: 3In the table, find the row with the email address ending with @cloudbuild.gserviceaccount.com
Select Edit principal
The roles to be granted to the Cloud Build service account are Compute Admin and Service Account User
If the required roles are not present, add them and click SAVE
В таблице найдите строку с адресом электронной почты, заканчивающимся на @cloudbuild.gserviceaccount.com
Выберите Править (Edit principal)
Роли, которые должны быть предоставлены учетной записи службы Cloud Build, - Сompute Admin и Service Account User
Если нужных ролей нет, добавьте и нажмите SAVE














Instructions for deploying MikoPBX in AWS via Terraform script
This guide describes deploying MikoPBX in AWS using the Infrastructure as Code (IaC) approach with Terraform. The entire infrastructure — EC2 instance, network rules, disks, and IP address — is declared in code, ensuring reproducibility, versioning, and the ability to quickly redeploy in any environment.
General process:
Download .raw → Upload to S3 → Import as AMI → Deploy via TerraformTerraform >= 1.3.0
AWS CLI configured with access keys (aws configure)
Bash (macOS / Linux)
IAM permissions: ec2:*, s3:*, iam:CreateRole, iam:PutRolePolicy
Configuring AWS CLI
Go to the MikoPBX releases page:
Download the latest image with the .raw extension.
Go to the .
Navigate to Services → Storage → S3.
Click Create bucket. Enter a unique bucket name in the Bucket name field. Use default values for all other fields.
Confirm by clicking Create bucket.
Open the created bucket by clicking its name. Click Upload and select the disk image file with the .raw extension.
Click Upload to confirm.
AWS requires a special IAM role vmimport for image imports. Perform these steps once per account.
Create a file trust-policy.json with the following content:
Create a file role-policy.json with the following content:
Replace
mikopbx-bucketwith your S3 bucket name.
Run the following commands to apply the policies:
Save the script below as import-image.sh and edit the variables DEFAULT_BUCKET, DEFAULT_IMAGE, and DEFAULT_NAME.
import-image.sh
Run the script:
Once complete, the script will output the AMI ID — save it, as it will be needed for Terraform.
Create all of the following files (directory structure):
Below we walk through each file and the content to add to each:
The main configuration file describing all AWS resources to be created: EC2 instance, Security Group, EBS disks, and Elastic IP. By default, the Security Group opens only the ports required for MikoPBX to operate: SSH, HTTP/HTTPS, SIP, and RTP.
Warning: Be sure to configure the Firewall in MikoPBX after your first login!
Declares variables with their types, descriptions, and default values. Does not contain specific values on its own — only the schema.
Defines what data Terraform will output after a successful apply: the web interface URL, and the login and password for the first login. Convenient for quickly retrieving credentials without opening the AWS Console.
Contains the specific variable values for your environment: region, AMI ID, instance type, etc. This is the file that changes when moving between environments (dev/staging/prod).
Note: Specify your own parameters in this file — replace
aws_region,instance_name,instance_type,storage_disk_size,allowed_ssh_cidr,create_key_pair, andpublic_key_pathas needed. Be sure to replacecustom_ami_idwith the ID of the AMI you created earlier.
Make sure all 4 files are created, then run the following commands:
You will see the following output:
Run the following command to preview the configuration:
You will see the configuration that Terraform has parsed and plans to create. Review all parameters, then run:
Enter yes to confirm. Upon successful creation of the MikoPBX instance, the required credentials will be displayed:
After a successful terraform apply:
Copy the URL from the output values.
Open it in your browser: https://<URL>
Use the credentials displayed during infrastructure creation to log in.
⚠️ After logging in, be sure to configure the Firewall in MikoPBX.
⚠️ The AMI and the S3 bucket containing the image are not deleted automatically — they must be removed manually via the AWS Console or CLI if no longer needed.
Error: InvalidAMIID.NotFound
Cause: The AMI exists in a different region.
Solution: Make sure the region in terraform.tfvars matches the region where the import script was run.
Error: OptInRequired during import
Cause: The vmimport role has not been created or lacks the required permissions.
Solution: Repeat the IAM role configuration step.
Error: import status error
Cause: Corrupted .raw file or incorrect format.
Solution: Verify that the original image was downloaded correctly and that the filename in DEFAULT_IMAGE is accurate.
Slow snapshot import
Importing a large image can take 10–30 minutes. The script automatically waits for completion, polling the status every 30 seconds.
aws configure
# AWS Access Key ID: your_key
# AWS Secret Access Key: your_secret_key
# Default region name: us-east-1 (your region)
# Default output format: json{
"Version": "2012-10-17",
"Statement": [
{
"Effect": "Allow",
"Principal": { "Service": "vmie.amazonaws.com" },
"Action": "sts:AssumeRole",
"Condition": {
"StringEquals": { "sts:Externalid": "vmimport" }
}
}
]
}{
"Version": "2012-10-17",
"Statement": [
{
"Effect": "Allow",
"Action": [
"s3:GetBucketLocation",
"s3:GetObject",
"s3:ListBucket"
],
"Resource": [
"arn:aws:s3:::mikopbx-bucket",
"arn:aws:s3:::mikopbx-bucket/*"
]
},
{
"Effect": "Allow",
"Action": [
"ec2:ModifySnapshotAttribute",
"ec2:CopySnapshot",
"ec2:RegisterImage",
"ec2:Describe*"
],
"Resource": "*"
}
]
}# Create the role
aws iam create-role \
--role-name vmimport \
--assume-role-policy-document "file://trust-policy.json"# Attach the policy to the role
aws iam put-role-policy \
--role-name vmimport \
--policy-name vmimport \
--policy-document "file://role-policy.json"#!/bin/bash
# -----------------------------------------------
# Settings — change these to match your values
# -----------------------------------------------
DEFAULT_IMAGE="mikopbx-2026.1.223-x86_64.raw"
DEFAULT_BUCKET="mikopbx-bucket"
DEFAULT_DESCRIPTION="MikoPBX PBX on Asterisk"
DEFAULT_NAME="MikoPBX 2026.1.223"
# Override via environment variables (optional)
IMAGE="${IMAGE:-$DEFAULT_IMAGE}"
BUCKET="${BUCKET:-$DEFAULT_BUCKET}"
DESCRIPTION="${DESCRIPTION:-$DEFAULT_DESCRIPTION}"
NAME="${NAME:-$DEFAULT_NAME}"
# -----------------------------------------------
# Import snapshot from S3
# -----------------------------------------------
JSON_FILE="disk_container.json"
cat <<EOF > ${JSON_FILE}
{
"Description": "${DESCRIPTION} image",
"Format": "raw",
"UserBucket": {
"S3Bucket": "${BUCKET}",
"S3Key": "${IMAGE}"
}
}
EOF
echo "Starting snapshot import..."
IMPORT_TASK_ID=$(aws ec2 import-snapshot \
--description "${DESCRIPTION} image" \
--disk-container "file://${JSON_FILE}" \
--query 'ImportTaskId' \
--output text)
echo "Import task started: $IMPORT_TASK_ID"
# Wait for import to complete
while true; do
STATUS=$(aws ec2 describe-import-snapshot-tasks \
--import-task-ids "$IMPORT_TASK_ID" \
--query 'ImportSnapshotTasks[0].SnapshotTaskDetail.Status' \
--output text)
echo "Status: $STATUS"
if [ "$STATUS" == "completed" ]; then
break
elif [ "$STATUS" == "error" ]; then
echo "Import failed!"
exit 1
fi
sleep 30
done
# Get snapshot ID
SNAPSHOT_ID=$(aws ec2 describe-import-snapshot-tasks \
--import-task-ids "$IMPORT_TASK_ID" \
--query 'ImportSnapshotTasks[0].SnapshotTaskDetail.SnapshotId' \
--output text)
echo "Snapshot created: $SNAPSHOT_ID"
# -----------------------------------------------
# Register AMI from snapshot
# -----------------------------------------------
AMI_ID=$(aws ec2 register-image \
--name "$NAME" \
--description "$DESCRIPTION" \
--architecture x86_64 \
--sriov-net-support simple \
--virtualization-type hvm \
--ena-support \
--boot-mode uefi \
--root-device-name "/dev/sda1" \
--block-device-mappings "[
{\"DeviceName\": \"/dev/sda1\", \"Ebs\": {
\"DeleteOnTermination\": true,
\"VolumeSize\": 1,
\"SnapshotId\": \"$SNAPSHOT_ID\"
}},
{\"DeviceName\": \"/dev/sdb\", \"Ebs\": {\"VolumeSize\": 50}}
]" \
--query 'ImageId' \
--output text)
echo ""
echo "=========================================="
echo "AMI successfully created: $AMI_ID"
echo "Use this ID in terraform.tfvars:"
echo " custom_ami_id = \"$AMI_ID\""
echo "=========================================="
# Remove temporary file
rm -f "$JSON_FILE"sh import-image.sh==========================================
AMI successfully created: ami-0c8820696110d0613
Use this ID in terraform.tfvars:
custom_ami_id = "ami-0c8820696110d0613"
==========================================mikopbx-aws-custom/
├── main.tf
├── variables.tf
├── outputs.tf
├── terraform.tfvarsterraform {
required_providers {
aws = {
source = "hashicorp/aws"
version = "~> 5.0"
}
}
required_version = ">= 1.3.0"
}
provider "aws" {
region = var.aws_region
}
# --------------------------------------------------
# Security Group
# --------------------------------------------------
resource "aws_security_group" "mikopbx_sg" {
name = "${var.instance_name}-sg"
description = "Security group for MikoPBX"
ingress {
from_port = 22
to_port = 22
protocol = "tcp"
cidr_blocks = [var.allowed_ssh_cidr]
description = "SSH"
}
ingress {
from_port = 443
to_port = 443
protocol = "tcp"
cidr_blocks = ["0.0.0.0/0"]
description = "HTTPS web UI"
}
ingress {
from_port = 80
to_port = 80
protocol = "tcp"
cidr_blocks = ["0.0.0.0/0"]
description = "HTTP"
}
ingress {
from_port = 5060
to_port = 5060
protocol = "udp"
cidr_blocks = ["0.0.0.0/0"]
description = "SIP UDP"
}
ingress {
from_port = 5060
to_port = 5060
protocol = "tcp"
cidr_blocks = ["0.0.0.0/0"]
description = "SIP TCP"
}
ingress {
from_port = 10000
to_port = 10200
protocol = "udp"
cidr_blocks = ["0.0.0.0/0"]
description = "RTP media"
}
egress {
from_port = 0
to_port = 0
protocol = "-1"
cidr_blocks = ["0.0.0.0/0"]
}
tags = {
Name = "${var.instance_name}-sg"
}
}
# --------------------------------------------------
# SSH Key Pair (optional)
# --------------------------------------------------
resource "aws_key_pair" "mikopbx_key" {
count = var.create_key_pair ? 1 : 0
key_name = "${var.instance_name}-key"
public_key = file(pathexpand(var.public_key_path))
}
# --------------------------------------------------
# EC2 Instance with custom AMI
# --------------------------------------------------
resource "aws_instance" "mikopbx" {
# Use the AMI created by import-image.sh
ami = var.custom_ami_id
instance_type = var.instance_type
key_name = var.create_key_pair ? aws_key_pair.mikopbx_key[0].key_name : var.existing_key_pair_name
vpc_security_group_ids = [aws_security_group.mikopbx_sg.id]
# System disk (1 GB)
root_block_device {
volume_size = 1
volume_type = "gp3"
}
tags = {
Name = var.instance_name
}
}
# --------------------------------------------------
# Disk for call recordings (50+ GB)
# --------------------------------------------------
resource "aws_ebs_volume" "mikopbx_storage" {
availability_zone = aws_instance.mikopbx.availability_zone
size = var.storage_disk_size
type = "gp3"
tags = {
Name = "${var.instance_name}-storage"
}
}
resource "aws_volume_attachment" "storage_attach" {
device_name = "/dev/sdc"
volume_id = aws_ebs_volume.mikopbx_storage.id
instance_id = aws_instance.mikopbx.id
}
# --------------------------------------------------
# Elastic IP
# --------------------------------------------------
resource "aws_eip" "mikopbx_eip" {
instance = aws_instance.mikopbx.id
domain = "vpc"
tags = {
Name = "${var.instance_name}-eip"
}
}variable "aws_region" {
description = "AWS region"
type = string
default = "us-east-1"
}
variable "custom_ami_id" {
description = "ID of the custom AMI created by import-image.sh"
type = string
# Value must be provided via terraform.tfvars
}
variable "instance_name" {
description = "EC2 instance name"
type = string
default = "mikopbx-vm"
}
variable "instance_type" {
description = "EC2 instance type"
type = string
default = "t3.micro"
}
variable "storage_disk_size" {
description = "Size of the recordings disk (GB)"
type = number
default = 50
}
variable "allowed_ssh_cidr" {
description = "CIDR block for SSH access"
type = string
default = "0.0.0.0/0"
}
variable "create_key_pair" {
description = "Create an SSH Key Pair (true) or use an existing one (false)"
type = bool
default = true
}
variable "public_key_path" {
description = "Path to the public SSH key"
type = string
default = "~/.ssh/id_rsa.pub"
}
variable "existing_key_pair_name" {
description = "Name of an existing Key Pair (if create_key_pair = false)"
type = string
default = ""
}output "first_login" {
description = "Credentials for the first login to the MikoPBX web interface"
value = <<-EOT
======================================
MikoPBX is ready!
======================================
URL: https://${aws_eip.mikopbx_eip.public_ip}
Login: admin
Password: ${aws_instance.mikopbx.id}
======================================
EOT
}aws_region = "ap-southeast-1"
custom_ami_id = "ami-0c8820696110d0613" # <- ID from import-image.sh
instance_name = "mikopbx-vm"
instance_type = "t3.micro"
storage_disk_size = 50
allowed_ssh_cidr = "0.0.0.0/0"
create_key_pair = true
public_key_path = "~/.ssh/id_rsa.pub"cd mikopbx-aws-custom # Navigate to the directory with the created files
terraform initTerraform has been successfully initialized!
You may now begin working with Terraform. Try running "terraform plan" to see
any changes that are required for your infrastructure. All Terraform commands
should now work.
If you ever set or change modules or backend configuration for Terraform,
rerun this command to reinitialize your working directory. If you forget, other
commands will detect it and remind you to do so if necessary.terraform planterraform apply======================================
MikoPBX is ready!
======================================
URL: https://52.221.99.139
Login: admin
Password: i-007352c23fa6d3b01
======================================terraform destroy# Delete the AMI
aws ec2 deregister-image --image-id ami-0a1b2c3d4e5f67890
# Delete the snapshot (ID can be found in AWS Console → EC2 → Snapshots)
aws ec2 delete-snapshot --snapshot-id snap-xxxxxxxxxxxxxxxxx
# Delete the file from S3
aws s3 rm s3://mikopbx-bucket/mikopbx-2026.1.223-x86_64.raw
# Delete the bucket (if empty)
aws s3 rb s3://mikopbx-bucketError: InvalidAMIID.NotFound: The image id 'ami-xxxx' does not existError: OptInRequiredStatus: error
Import failed!




Instructions with examples on creating and using API keys
Working with the REST API follows the OpenAPI standard. To get the current list of endpoints, use the "Documentation" section inside the PBX. Below are examples of working with the main features of the MikoPBX REST interface.
To run all examples in this guide, create an API key and configure the following access permissions (see the general article for details):
In this article we will be working with Python, so you need to install the required dependencies:
Below is a connection template for accessing the station via an API key. Use it before all scripts in this guide. The API key is passed directly in the request header — no additional authentication is required:
Endpoint: POST /pbxcore/api/v3/employees
The table below lists the parameters for this request.
Example API response (HTTP 201):
Possible response codes:
On successful execution, you will see the following console output:
Employees 283 and 284 will be created on the station.
On successful execution, you will see the following console output:
On successful execution, you will see the following console output:
3 employees will be created on the station.
Endpoint: POST /pbxcore/api/v3/sip-providers
On successful execution, you will see the following console output:
A provider will be created on the station:
On successful execution, you will see the following console output:
Endpoint: GET /pbxcore/api/v3/cdr — read-only.
On successful execution, you will see the following console output:
On successful execution, you will see the following console output:
Endpoints: GET /pbxcore/api/v3/sip , GET /pbxcore/api/v3/sip-providers
On successful execution, you will see the following console output:
Employee statuses (status field)
Provider statuses (state field)
Endpoint: GET /pbxcore/api/v3/pbx-status
On successful execution, you will see the following console output:
The full list of endpoints and interactive documentation is available in the section.
Employees Management
Read and write
Creating and editing employees
Providers
Read and write
Creating and editing providers
SIP
Read
Employee and trunk registration statuses
Call Records
Read
Call history (CDR)
PBX Status
Read
Active calls in real time
SIP Providers
Read and write
Creating and editing SIP providers
pip install requestsimport requests
BASE_URL = 'https://your-mikopbx.com/pbxcore/api/v3'
API_KEY = 'your-api-key'
HEADERS = {
'Authorization': f'Bearer {API_KEY}',
'Content-Type': 'application/json',
}number
✅
string, 2–8 digits
Extension number
def create_employee(
number: str,
name: str,
sip_secret: str,
email: str = '',
mobile: str = '',
record_calls: bool = True,
fwd_ringlength: int = 45,
) -> dict:
payload = {
'number': number,
'user_username': name,
'sip_secret': sip_secret,
'sip_enableRecording': record_calls,
'fwd_ringlength': fwd_ringlength,
}
if email: payload['user_email'] = email
if mobile: payload['mobile_number'] = mobile
r = requests.post(f'{BASE_URL}/employees', headers=HEADERS, json=payload)
result = r.json()
if result.get('result'):
print(f" Created: {number} ({name}), id={result['data']['id']}")
else:
print(f" Error: {result.get('messages', {}).get('error', [])}")
return result
# Minimal example (required fields only)
create_employee(
number='243',
name='John Smith',
sip_secret='Secure#Pass9201',
)
# Full example
create_employee(
number='244',
name='Anna Johnson',
sip_secret='Secure#Pass9202',
email='anna@company.com',
mobile='79001234567',
record_calls=True,
fwd_ringlength=30,
){
"result": true,
"data": {
"number": "201",
"user_username": "John Smith",
"sip_secret": "Secure#Pass9201",
"sip_dtmfmode": "auto",
"sip_transport": "udp",
"sip_enableRecording": true,
"sip_networkfilterid": "none",
"fwd_ringlength": 45,
"id": "1",
"extensions_length": 3
},
"messages": {"error": [], "info": [], "warning": []}
}201
Employee successfully created
400
Validation error (weak password <5 characters, invalid number format)
401
Created: 283 (John Smith), id=113
Created: 284 (Anna Johnson), id=114
Process finished with exit code 0def list_employees(search: str = '', limit: int = 100, offset: int = 0) -> list:
params = {'limit': limit, 'offset': offset}
if search: params['search'] = search
r = requests.get(f'{BASE_URL}/employees', headers=HEADERS, params=params)
return r.json().get('data', {}).get('data', [])
for emp in list_employees():
print(f" {emp.get('number'):>6} {emp.get('user_username', '')}") 202 Brown Brandon
203 Collins Melanie
201 Smith James
283 John Smith
284 Anna Johnson
Process finished with exit code 0import time
employees = [
{'number': '291', 'name': 'John Smith', 'secret': 'Pass#9201'},
{'number': '292', 'name': 'Anna Johnson', 'secret': 'Pass#9202'},
{'number': '293', 'name': 'Peter Brown', 'secret': 'Pass#9203'},
]
created, failed = [], []
for emp in employees:
r = requests.post(
f'{BASE_URL}/employees',
headers=HEADERS,
json={
'number': emp['number'],
'user_username': emp['name'],
'sip_secret': emp['secret'],
}
)
result = r.json()
if result.get('result'):
created.append(emp['number'])
print(f" {emp['number']} {emp['name']}")
else:
failed.append(emp['number'])
print(f" {emp['number']}: {result.get('messages', {}).get('error', [])}")
time.sleep(0.2) # small pause between requests
print(f'Created: {len(created)}, Errors: {len(failed)}') 291 John Smith
292 Anna Johnson
293 Peter Brown
Created: 3, Errors: 0
Process finished with exit code 0description
✅
string
Provider name
def create_sip_provider(
description: str,
host: str,
username: str = '',
password: str = '',
registration_type: str = 'outbound',
qualify: bool = True,
) -> dict:
payload = {
'description': description,
'host': host,
}
if username: payload['username'] = username
if password: payload['secret'] = password
if registration_type: payload['registration_type'] = registration_type
if not qualify: payload['qualify'] = qualify
r = requests.post(f'{BASE_URL}/sip-providers', headers=HEADERS, json=payload)
result = r.json()
if result.get('result'):
print(f" Provider created: {description}")
else:
print(f" Error: {result.get('messages', {}).get('error', [])}")
return result
create_sip_provider(
description='Zadarma',
host='sip.zadarma.com',
username='316811',
password='mysecretpass',
) Provider created: Zadarma
Process finished with exit code 0def list_providers() -> list:
r = requests.get(f'{BASE_URL}/sip-providers', headers=HEADERS)
return r.json().get('data', [])
for prov in list_providers():
print(f" {prov.get('id'):<20} {prov.get('description', '')} [{prov.get('type', '')}]") SIP-TRUNK-34F7CAFE [SIP]
SIP-TRUNK-7B5977ED [SIP]
Process finished with exit code 0offset
integer
Pagination offset (default: 0)
limit
integer
from datetime import datetime, timedelta
def get_cdr(
offset: int = 0,
limit: int = 20,
date_from: str = None,
date_to: str = None,
src_num: str = None,
dst_num: str = None,
disposition: str = None,
) -> list:
params = {'offset': offset, 'limit': min(limit, 100)}
if date_from: params['dateFrom'] = date_from
if date_to: params['dateTo'] = date_to
if src_num: params['src_num'] = src_num
if dst_num: params['dst_num'] = dst_num
if disposition: params['disposition'] = disposition
r = requests.get(f'{BASE_URL}/cdr', headers=HEADERS, params=params)
return r.json().get('data', {}).get('records', [])
now = datetime.now()
then = now - timedelta(days=7)
for row in get_cdr(
date_from=then.strftime('%Y-%m-%dT%H:%M:%S'),
date_to=now.strftime('%Y-%m-%dT%H:%M:%S'),
):
print(
str(row.get('start', ''))[:16],
row.get('src_num', ''), '→', row.get('dst_num', ''),
row.get('disposition', ''), row.get('totalBillsec', 0), 's'
)2026-03-17 13:30 252 → 202 ANSWERED 48 s
2026-03-17 13:30 243 → 252 BUSY 0 s
2026-03-17 13:30 243 → 89161111111 CHANUNAVAIL 0 s
2026-03-17 13:29 202 → 243 NOANSWER 0 s
2026-03-17 13:29 202 → 202 ANSWERED 2 s
2026-03-17 13:29 202 → 243 NOANSWER 0 s
2026-03-17 13:29 202 → 10003246 NOANSWER 0 s
2026-03-17 13:28 202 → 243 NOANSWER 0 s
Process finished with exit code 0def cdr_stats(days: int = 1) -> dict:
now = datetime.now()
then = now - timedelta(days=days)
records = get_cdr(
date_from=then.strftime('%Y-%m-%dT%H:%M:%S'),
date_to=now.strftime('%Y-%m-%dT%H:%M:%S'),
limit=100
)
answered = [r for r in records if r.get('disposition') == 'ANSWERED']
missed = [r for r in records if r.get('disposition') in ('NO ANSWER', 'NOANSWER')]
total_dur = sum(r.get('totalBillsec', 0) for r in answered)
return {
'total': len(records),
'answered': len(answered),
'missed': len(missed),
'avg_sec': total_dur // len(answered) if answered else 0,
}
stats = cdr_stats(days=7)
print(f"Calls over 7 days: {stats['total']}")
print(f"Answered: {stats['answered']}")
print(f"Missed: {stats['missed']}")
print(f"Avg. duration: {stats['avg_sec']}s")Calls over 7 days: 13
Answered: 2
Missed: 5
Avg. duration: 25s
Process finished with exit code 0linkedid
string
Unique call identifier
start
datetime
from datetime import datetime
def show_employees():
r = requests.get(f'{BASE_URL}/sip:getStatuses', headers=HEADERS)
peers = r.json().get('data', {})
for number, info in peers.items():
icon = '🟢' if info.get('status') == 'Available' else '🔴'
print(f" {icon} {number:>6} {info.get('callerid', '')} [{info.get('status', '')}]")
def show_providers():
r = requests.get(f'{BASE_URL}/sip-providers:getStatuses', headers=HEADERS)
providers = r.json().get('data', {}).get('sip', {})
for prov_id, info in providers.items():
icon = '🟢' if info.get('state') == 'registered' else '🔴'
print(f" {icon} {info.get('description', prov_id):>20} {info.get('username', '')}@{info.get('host', '')} [{info.get('state', '')}]")
if __name__ == '__main__':
print(f'MikoPBX Monitor [{datetime.now().strftime("%Y-%m-%d %H:%M:%S")}]')
print('\n── Employees ───────────────────────────────')
show_employees()
print('\n── Providers ───────────────────────────────')
show_providers()MikoPBX Monitor [2026-03-17 16:47:35]
── Employees ───────────────────────────────
🔴 201 Smith James [Unavailable]
🟢 202 Brown Brandon [Available]
🔴 203 Collins Melanie [Unavailable]
🔴 243 John Smith [Unavailable]
🟢 244 Anna Johnson [Available]
🔴 251 John Smith [Unavailable]
🟢 252 Anna Johnson [Available]
🔴 253 Peter Brown [Unavailable]
── Providers ───────────────────────────────
🔴 Demo provider SIP-PROVIDER-122642725b9265fd7151c@demo.askozia.ru [rejected]
🟢 Zadarma 316811@sip.zadarma.com [registered]
Process finished with exit code 0Available
Registered and available
Unavailable
Not registered (offline)
registered
Registered on the provider's server
rejected
Registration rejected by the server
unregistered
def get_active_calls() -> list:
r = requests.get(f'{BASE_URL}/pbx-status:getActiveCalls', headers=HEADERS)
return r.json().get('data', [])
calls = get_active_calls()
print(f'Active calls: {len(calls)}')
for call in calls:
print(f" {call.get('src_num', '?')} → {call.get('dst_num', '?')} [{call.get('src_name', '')} → {call.get('dst_name', '')}]")Active calls: 1
243 → 252 [John Smith → Anna Johnson]
Process finished with exit code 0import urllib3
urllib3.disable_warnings()user_username
✅
string, 1–100 characters
Employee full name
sip_secret
✅
string, 5–100 characters
SIP account password
user_email
—
string email, ≤255
Email for notifications
mobile_number
—
string E.164, ≤50
Mobile number (+1...) for forwarding
mobile_dialstring
—
string, ≤255
Mobile dial string
sip_transport
—
udp / tcp / tls / udp,tcp
SIP transport (default: udp)
sip_dtmfmode
—
auto / rfc4733 / inband / info
DTMF mode (default: auto)
sip_enableRecording
—
boolean
Call recording (default: true)
sip_networkfilterid
—
number | "none"
Network filter ID
sip_manualattributes
—
string, ≤1024
Additional SIP parameters
fwd_ringlength
—
integer, ≤180
Ring time before forwarding (sec, default: 45)
fwd_forwarding
—
number | hangup | busy
Unconditional forwarding
fwd_forwardingonbusy
—
number | hangup | busy
Forwarding on busy
fwd_forwardingonunavailable
—
number | hangup | busy
Forwarding on unavailable
Invalid or missing API key
403
No write permission for the /employees resource
409
Conflict — number already in use
host
✅
string
Provider SIP server address
username
—
string
Login on the provider's server
secret
—
string
Password
registration_type
—
string
inbound / outbound / none
qualify
—
boolean
Availability monitoring (default: true)
transport
—
string
udp / tcp / tls / udp,tcp (default: udp,tcp)
dtmfmode
—
string
auto / rfc4733 / inband / info (default: auto)
port
—
integer
Connection port (default: 5060)
disabled
—
boolean
Disable provider (default: false)
Number of records, max. 100
dateFrom
string
Period start: %Y-%m-%dT%H:%M:%S
dateTo
string
Period end: %Y-%m-%dT%H:%M:%S
src_num
string
Filter by caller number
dst_num
string
Filter by destination number
disposition
string
ANSWERED / NO ANSWER / BUSY / FAILED
Call start time
src_num
string
Caller number
src_name
string
Caller name
dst_num
string
Destination number
dst_name
string
Destination name
disposition
string
ANSWERED / NO ANSWER / NOANSWER / BUSY / CHANUNAVAIL / FAILED
totalBillsec
integer
Call duration (seconds)
totalDuration
integer
Total duration (including ringing)
records
array
Detailed records for each call leg
records[].recordingfile
string
Path to the recording file
records[].playback_url
string
URL for playing back the recording
records[].download_url
string
URL for downloading the recording
records[].dtmf_digits
string
DTMF digits pressed in IVR
Not registered




Description of documentation and endpoint table for working with REST API in MikoPBX
MikoPBX REST API follows the OpenAPI standard. The interactive documentation is built directly into the PBX and always contains the current list of endpoints, parameters, and schemas for your version of the system.
Go to "System" → "API Keys".
Click the "API Documentation" button.
The documentation is built on the OpenAPI standard and provides a complete description of all MikoPBX REST API endpoints.
Endpoint navigation — in the left panel, all endpoints are grouped by section.
For each endpoint, a brief description is shown along with the request method (GET, POST, PUT, PATCH, DELETE) and the endpoint with the PBX address substituted. All available request parameters are displayed below.
Code examples — ready-made request examples in different languages are available for each endpoint. The switcher is located below the parameters panel — Shell / cURL is shown by default, other languages are also available (click the language name to switch — in this guide, Python 3).
A server response example is shown below.
Online request execution — the documentation allows you to send real requests directly from the browser and receive responses from your PBX. The server is determined automatically from the current page address.
At the bottom of the page you will find possible response codes with brief explanations, as well as all body parameters for the selected response.
Base prefix for all paths:
/pbxcore/api/v3
— exports ban decisions for external bouncers (CrowdSec-compatible). Available starting from MikoPBX 2026.1.76.
GET
/employees
Get list of employees
POST
/employees
GET
/extensions
Get list of extensions
GET
/extensions/{id}
GET
/sip-providers
Get list of SIP providers
POST
/sip-providers
GET
/iax-providers
Get list of IAX providers
POST
/iax-providers
GET
/providers
Get list of all providers
GET
/providers/{id}
GET
/call-queues
Get list of queues
POST
/call-queues
GET
/ivr-menu
Get list of IVR menus
POST
/ivr-menu
GET
/incoming-routes
Get list of incoming routes
POST
/incoming-routes
GET
/outbound-routes
Get list of outbound routes
POST
/outbound-routes
GET
/off-work-times
Get list of time conditions
POST
/off-work-times
GET
/conference-rooms
Get list of conference rooms
POST
/conference-rooms
GET
/dialplan-applications
Get list of dialplan applications
POST
/dialplan-applications
GET
/sound-files
Get list of sound files
POST
/sound-files
GET
/custom-files
Get list of custom files
POST
/custom-files
GET
/pbx-status:getActiveCalls
Get active calls
GET
/pbx-status:getActiveChannels
GET
/sip:getStatuses
Get statuses of all SIP devices
GET
/sip:getPeersStatuses
GET
/sip-providers:getStatuses
Get statuses of all SIP providers
GET
/sip-providers/{id}:getStatus
GET
/iax-providers:getStatuses
Get statuses of all IAX providers
GET
/iax-providers/{id}:getStatus
GET
/providers:getStatuses
Get statuses of all providers
GET
/providers/{id}:getStatus
GET
/cdr
Get list of CDR records
GET
/cdr/{id}
GET
/advice:getList
Get list of system notifications
GET
/advice:refresh
POST
/auth:login
Log in to the system
POST
/auth:refresh
GET
/api-keys
Get list of API keys
POST
/api-keys
GET
/asterisk-managers
Get list of AMI users
POST
/asterisk-managers
GET
/asterisk-rest-users
Get list of ARI users
POST
/asterisk-rest-users
GET
/passkeys
Get list of passkeys
POST
/passkeys
GET
/passwords:generate
Generate password
POST
/passwords:validate
GET
/users:available
Check email availability
GET
/network-filters
Get list of network filters
GET
/network-filters/{id}
GET
/system:ping
Check system availability
GET
GET
/general-settings
Get general settings
PUT
/general-settings
GET
/network
Get list of network interfaces
GET
/network/{id}
GET
/firewall
Get list of firewall rules
POST
/firewall
GET
/fail2ban
Get Fail2Ban settings
PUT
/fail2ban
GET
/time-settings:getAvailableTimezones
Get list of available timezones
GET
/mail-settings
Get mail settings
PUT
/mail-settings
GET
/storage:usage
Get storage usage statistics
GET
/storage:list
GET
/s3-storage
Get S3 storage configuration
PUT
/s3-storage
GET
/modules
Get list of modules
POST
GET
/license:getLicenseInfo
Get license information
GET
/license:ping
GET
/files/{id}
Get file contents
PUT
/files/{id}
GET
/sysinfo:getInfo
Get system information
GET
/sysinfo:getExternalIpInfo
GET
/syslog:getLogsList
Get list of log files
POST
/syslog:getLogFromFile
GET
/openapi:getSpecification
Get OpenAPI specification
GET
/openapi:getAclRules
GET
/search:getSearchItems
Global search
GET
/wiki-links:getLink
Get documentation link
POST
/user-page-tracker:pageView
Track page view
POST
/user-page-tracker:pageLeave


Create a new employee
GET
/employees/{id}
Get employee by ID
PUT
/employees/{id}
Update employee
PATCH
/employees/{id}
Partially update employee
DELETE
/employees/{id}
Delete employee
GET
/employees:getDefault
Get default values
POST
/employees:batchCreate
Batch create employees
POST
/employees:batchDelete
Batch delete employees
POST
/employees:import
Import employees (preview)
POST
/employees:confirmImport
Confirm import
POST
/employees:export
Export employees
POST
/employees:exportTemplate
Export template
Get extension by ID
POST
/extensions:available
Check number availability
GET
/extensions:getForSelect
Get extensions for dropdown list
POST
/extensions/{id}:getPhoneRepresent
Get phone representation
POST
/extensions:getPhonesRepresent
Get phones representation
Create SIP provider
GET
/sip-providers/{id}
Get SIP provider by ID
PUT
/sip-providers/{id}
Update SIP provider
PATCH
/sip-providers/{id}
Partially update SIP provider
DELETE
/sip-providers/{id}
Delete SIP provider
GET
/sip-providers/{id}:copy
Copy SIP provider
GET
/sip-providers:getDefault
Get SIP provider template
Create IAX provider
GET
/iax-providers/{id}
Get IAX provider by ID
PUT
/iax-providers/{id}
Update IAX provider
PATCH
/iax-providers/{id}
Partially update IAX provider
DELETE
/iax-providers/{id}
Delete IAX provider
GET
/iax-providers/{id}:copy
Copy IAX provider
GET
/iax-providers:getDefault
Get IAX provider template
Get provider by ID
GET
/providers:getForSelect
Get providers for dropdown list
Create a new queue
GET
/call-queues/{id}
Get queue by ID
PUT
/call-queues/{id}
Update queue
PATCH
/call-queues/{id}
Partially update queue
DELETE
/call-queues/{id}
Delete queue
GET
/call-queues/{id}:copy
Copy queue
GET
/call-queues:getDefault
Get default values
Create a new IVR menu
GET
/ivr-menu/{id}
Get IVR menu by ID
PUT
/ivr-menu/{id}
Update IVR menu
PATCH
/ivr-menu/{id}
Partially update IVR menu
DELETE
/ivr-menu/{id}
Delete IVR menu
GET
/ivr-menu/{id}:copy
Copy IVR menu
GET
/ivr-menu:getDefault
Get default values
Create incoming route
GET
/incoming-routes/{id}
Get incoming route by ID
PUT
/incoming-routes/{id}
Update incoming route
PATCH
/incoming-routes/{id}
Partially update incoming route
DELETE
/incoming-routes/{id}
Delete incoming route
POST
/incoming-routes/{id}:copy
Copy incoming route
GET
/incoming-routes:getDefault
Get default values
GET
/incoming-routes:getDefaultRoute
Get default route
POST
/incoming-routes:changePriority
Change route priorities
Create outbound route
GET
/outbound-routes/{id}
Get outbound route by ID
PUT
/outbound-routes/{id}
Update outbound route
PATCH
/outbound-routes/{id}
Partially update outbound route
DELETE
/outbound-routes/{id}
Delete outbound route
GET
/outbound-routes/{id}:copy
Copy outbound route
GET
/outbound-routes:getDefault
Get default values
POST
/outbound-routes:changePriority
Change route priorities
Create time condition
GET
/off-work-times/{id}
Get time condition by ID
PUT
/off-work-times/{id}
Update time condition
PATCH
/off-work-times/{id}
Partially update time condition
DELETE
/off-work-times/{id}
Delete time condition
GET
/off-work-times/{id}:copy
Copy time condition
GET
/off-work-times:getDefault
Get default values
POST
/off-work-times:changePriorities
Change condition priorities
Create conference room
GET
/conference-rooms/{id}
Get conference room by ID
PUT
/conference-rooms/{id}
Update conference room
PATCH
/conference-rooms/{id}
Partially update conference room
DELETE
/conference-rooms/{id}
Delete conference room
GET
/conference-rooms:getDefault
Get conference room template
Create dialplan application
GET
/dialplan-applications/{id}
Get dialplan application by ID
PUT
/dialplan-applications/{id}
Update dialplan application
PATCH
/dialplan-applications/{id}
Partially update dialplan application
DELETE
/dialplan-applications/{id}
Delete dialplan application
GET
/dialplan-applications/{id}:copy
Copy dialplan application
GET
/dialplan-applications:getDefault
Get dialplan application template
Create sound file
GET
/sound-files/{id}
Get sound file by ID
PUT
/sound-files/{id}
Update sound file
PATCH
/sound-files/{id}
Partially update sound file
DELETE
/sound-files/{id}
Delete sound file
GET
/sound-files:getDefault
Get default values
GET
/sound-files:getForSelect
Get for dropdown list
GET
/sound-files:playback
Play back sound file
POST
/sound-files:uploadFile
Upload sound file
POST
/sound-files:convertAudioFile
Convert audio file
Create new custom file
GET
/custom-files/{id}
Get custom file by ID
PUT
/custom-files/{id}
Update custom file
PATCH
/custom-files/{id}
Partially update custom file
DELETE
/custom-files/{id}
Delete custom file
GET
/custom-files:getDefault
Get default values
Get active channels
Get SIP peer statuses (legacy)
GET
/sip:getRegistry
Get registration status (legacy)
POST
/sip:processAuthFailures
Process authentication failures
GET
/sip/{id}:getStatus
Get SIP device status
GET
/sip/{id}:getStats
Get SIP device statistics
GET
/sip/{id}:getHistory
Get connection history
GET
/sip/{id}:getSecret
Get SIP password
GET
/sip/{id}:getAuthFailureStats
Get authentication failure statistics
POST
/sip/{id}:clearAuthFailureStats
Clear failure statistics
POST
/sip/{id}:forceCheck
Force status check
Get SIP provider status
GET
/sip-providers/{id}:getHistory
Get connection history
GET
/sip-providers/{id}:getStats
Get SIP provider statistics
POST
/sip-providers/{id}:forceCheck
Force registration check
POST
/sip-providers/{id}:updateStatus
Update provider status
Get IAX provider status
GET
/iax-providers/{id}:getHistory
Get connection history
GET
/iax-providers/{id}:getStats
Get IAX provider statistics
POST
/iax-providers/{id}:forceCheck
Force registration check
POST
/iax-providers/{id}:updateStatus
Update provider status
GET
/iax:getRegistry
Get IAX provider registration status
Get provider status
GET
/providers/{id}:getHistory
Get provider history
GET
/providers/{id}:getStats
Get provider statistics
POST
/providers/{id}:updateStatus
Update provider status
Get CDR record by ID
DELETE
/cdr/{id}
Delete CDR record
GET
/cdr:getMetadata
Get CDR metadata
GET
/cdr:playback
Play back call recording
GET
/cdr:download
Download call recording
Refresh notification cache
Refresh access token
POST
/auth:logout
Log out of the system
Create a new API key
GET
/api-keys/{id}
Get API key by ID
PUT
/api-keys/{id}
Update API key
PATCH
/api-keys/{id}
Partially update API key
DELETE
/api-keys/{id}
Delete API key
GET
/api-keys:getDefault
Get default values
POST
/api-keys:generateKey
Generate a new key
Create a new AMI user
GET
/asterisk-managers/{id}
Get AMI user by ID
PUT
/asterisk-managers/{id}
Update AMI user
PATCH
/asterisk-managers/{id}
Partially update AMI user
DELETE
/asterisk-managers/{id}
Delete AMI user
GET
/asterisk-managers/{id}:copy
Copy AMI user
GET
/asterisk-managers:getDefault
Get default values
Create a new ARI user
GET
/asterisk-rest-users/{id}
Get ARI user by ID
PUT
/asterisk-rest-users/{id}
Update ARI user
PATCH
/asterisk-rest-users/{id}
Partially update ARI user
DELETE
/asterisk-rest-users/{id}
Delete ARI user
GET
/asterisk-rest-users:getDefault
Get default values
Create a new passkey
GET
/passkeys/{id}
Get passkey by ID
PATCH
/passkeys/{id}
Update passkey
DELETE
/passkeys/{id}
Delete passkey
GET
/passkeys:checkAvailability
Check passkey availability
GET
/passkeys:authenticationStart
Start passkey authentication
POST
/passkeys:authenticationFinish
Finish passkey authentication
POST
/passkeys:registrationStart
Start passkey registration
POST
/passkeys:registrationFinish
Finish passkey registration
Validate password strength
POST
/passwords:checkDictionary
Check password against dictionary
POST
/passwords:batchValidate
Batch validate passwords
POST
/passwords:batchCheckDictionary
Batch dictionary check
Get network filter by ID
GET
/network-filters:getForSelect
Get filters for dropdown list
/system:checkAuth
Check authentication
GET
/system:datetime
Get system time
GET
/system:getAvailableLanguages
Get available languages
GET
/system:checkForUpdates
Get detailed update information
GET
/system:checkIfNewReleaseAvailable
Quick check for new version availability
GET
/system:getDeleteStatistics
Get deletion statistics
POST
/system:reboot
Reboot the system
POST
/system:shutdown
Shut down the system
POST
/system:upgrade
Upgrade the system
POST
/system:restoreDefault
Restore default settings
POST
/system:changeLanguage
Change system language
POST
/system:convertAudioFile
Convert audio file
POST
/system:executeBashCommand
Execute bash command
POST
/system:executeSqlRequest
Execute SQL query
POST
/system:updateMailSettings
Update mail settings
Update general settings
PATCH
/general-settings
Partially update general settings
GET
/general-settings/{id}
Get specific setting
GET
/general-settings:getDefault
Get default values
POST
/general-settings:updateCodecs
Update codec settings
Get network interface by ID
DELETE
/network/{id}
Delete network interface
GET
/network:getConfig
Get full network configuration
GET
/network:getNatSettings
Get NAT settings
POST
/network:saveConfig
Save network configuration
Create firewall rule
GET
/firewall/{id}
Get firewall rule by ID
PUT
/firewall/{id}
Update firewall rule
PATCH
/firewall/{id}
Partially update firewall rule
DELETE
/firewall/{id}
Delete firewall rule
GET
/firewall:getDefault
Get default values
GET
/firewall:getBannedIps
Get list of banned IPs
POST
/firewall:unbanIp
Unban IP address
POST
/firewall:enable
Enable firewall
POST
/firewall:disable
Disable firewall
Update Fail2Ban settings
PATCH
/fail2ban
Partially update Fail2Ban settings
Update mail settings
PATCH
/mail-settings
Partially update mail settings
DELETE
/mail-settings
Reset mail settings
GET
/mail-settings:getDefault
Get default values
GET
/mail-settings:getDiagnostics
Get mail settings diagnostics
GET
/mail-settings:getOAuth2Url
Get OAuth2 authorization URL
POST
/mail-settings:refreshToken
Refresh OAuth2 token
POST
/mail-settings:testConnection
Test SMTP server connection
POST
/mail-settings:sendTestEmail
Send test email
Get list of available storage devices
POST
/storage:mount
Mount storage device
POST
/storage:umount
Unmount storage device
POST
/storage:mkfs
Format storage device
POST
/storage:statusMkfs
Get formatting status
Update S3 storage configuration
PATCH
/s3-storage
Partially update S3 configuration
GET
/s3-storage:stats
Get S3 synchronization statistics
GET
/s3-storage:testConnection
Test S3 connection
/modules
Create module
GET
/modules/{id}
Get module by ID
PUT
/modules/{id}
Update module
PATCH
/modules/{id}
Partially update module
DELETE
/modules/{id}
Delete module
GET
/modules/{id}:getModuleInfo
Get module information
GET
/modules/{id}:getModuleLink
Get module download link
GET
/modules/{id}:getDownloadStatus
Get download status
POST
/modules/{id}:startDownload
Start module download
POST
/modules/{id}:installFromRepo
Install module from repository
POST
/modules:installFromPackage
Install module from package
POST
/modules:getMetadataFromPackage
Get metadata from package
POST
/modules/{id}:enable
Enable module
POST
/modules/{id}:disable
Disable module
POST
/modules/{id}:uninstall
Uninstall module
POST
/modules:updateAll
Update all modules
GET
/modules:getAvailableModules
Get available modules
GET
/modules:getInstallationStatus
Get installation status
GET
/modules:getDefault
Get default module settings
Check license server connection
GET
/license:resetKey
Reset license key
GET
/license:sendPBXMetrics
Send PBX metrics
POST
/license:captureFeatureForProductId
Capture feature for product
POST
/license:processUserRequest
Process user request
Upload/update file
DELETE
/files/{id}
Delete file
POST
/files:upload
Upload file (chunked)
GET
/files:uploadStatus
Check upload status
POST
/files:downloadFirmware
Download firmware
GET
/files:firmwareStatus
Check firmware download status
Get external IP address
GET
/sysinfo:getHypervisorInfo
Get hypervisor information
GET
/sysinfo:getDMIInfo
Get DMI information
Get log file contents
POST
/syslog:getLogTimeRange
Get log time range
POST
/syslog:eraseFile
Clear log file
POST
/syslog:startCapture
Start packet capture
POST
/syslog:stopCapture
Stop packet capture
POST
/syslog:prepareArchive
Prepare log archive
POST
/syslog:downloadArchive
Download log archive
POST
/syslog:downloadLogFile
Download log file
Get API ACL rules
GET
/openapi:getDetailedPermissions
Get detailed permissions list
GET
/openapi:getSimplifiedPermissions
Get simplified permissions list
GET
/openapi:getValidationSchemas
Get validation schemas
POST
/openapi:clearCache
Clear OpenAPI cache
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