In the world of VoIP, choosing the right PBX platform is crucial for service providers aiming to deliver reliable, scalable, and feature-rich communication solutions. With a variety of PBX platforms available, each offering unique capabilities, VoIP providers can tailor their services to meet specific client needs. In this post, we explore the top PBX platforms, highlighting their strengths and features to help you make an informed choice for your VoIP business.

PBX Functions Addon in MOR

If you’re using (or planning to use) MOR—a Class 5 Softswitch with integrated billing, routing, rate management, and reporting—you can enhance its capabilities with the PBX Functions Addon. This add-on provides essential PBX features, including External DID, Ring Groups, Voicemail, PhoneBook, Music on Hold, Call Queues, IVR System, Call Flow, BLF, and PBX Pools. For those needing more advanced functionalities, MOR supports integration with third-party PBX systems. Any SIP-compatible PBX can work with MOR, but if you’re unsure which option is best, here are some of the most popular choices among our clients:

1. Asterisk

Asterisk is an open-source PBX platform that has been a backbone of VoIP for over two decades. Known for its flexibility and vast array of customizable features, Asterisk is highly favored among developers and businesses that require bespoke solutions. With its extensive APIs and integration capabilities, Asterisk serves as the foundation for several PBX systems, including FreePBX, VitalPBX and iVozProvider.

2. FreePBX

FreePBX builds upon Asterisk by providing a user-friendly, web-based interface that simplifies management and configuration. Ideal for businesses needing an open-source solution with reliable community support, FreePBX offers extensive call routing, voicemail, IVR, and conferencing features. It’s widely used by small to medium businesses due to its ease of use and scalability.

3. PBXact

PBXact is essentially FreePBX with commercial enhancements and premium support offered by Sangoma. With the option for perpetual licensing, PBXact provides added stability and features over FreePBX, making it suitable for enterprises and service providers looking for reliability and professional support. It’s an ideal solution for those who need commercial-grade capabilities without leaving the FreePBX ecosystem.

4. 3CX

3CX is a software-based PBX known for its user-friendly experience, unified communications capabilities, and support for both on-premises and cloud-based deployments. With its comprehensive feature set, 3CX caters to small, medium, and large enterprises, offering features such as video conferencing, web chat, and CRM integration. Its multi-platform support on Windows, Linux, and Android/iOS makes it a versatile choice.

You can find detailed guides on connecting 3CX to MOR and M4 in the Kolmisoft Wiki. Additionally, some Kolmisoft clients are listed on the 3CX Supported SIP Trunk Providers page. Being recognized as a 3CX-supported SIP Trunk Provider significantly enhances your market visibility, positioning you as a trusted choice for businesses seeking reliable VoIP solutions. However, there is also a risk of losing this status, which would impact an important marketing channel. For more details on such cases, refer to this blog post: 3CX removed us from the Supported SIP Trunk Providers page.

5. 2600hz Kazoo

Kazoo by 2600hz is a multi-tenant, carrier-grade PBX that is unique in its non-instance-based architecture. Kazoo is designed for large-scale VoIP deployments, providing robust APIs that allow service providers to build custom applications and automation workflows. Ideal for service providers seeking scalability and flexibility, Kazoo stands out with its focus on carrier-grade multi-tenancy.

We have previously implemented Kazoo for several Kolmisoft clients and have been generally satisfied with the platform’s performance. You can find the Online Demo of Kazoo Platform in the Kolmisoft Wiki.

6. FusionPBX

FusionPBX is a web interface for FreeSWITCH, offering a multi-tenant PBX solution with an extensive list of features, from call routing to IVR. FusionPBX is well-suited for service providers who prefer FreeSWITCH as their underlying technology and need a reliable, customizable multi-tenant environment.

7. VitalPBX

VitalPBX provides a modern and intuitive PBX experience built on Asterisk. Known for its extensive feature set and professional-grade interface, VitalPBX targets service providers and enterprise environments that need advanced functionalities, such as multi-tenancy, billing, and custom branding options. It offers a balanced combination of user-friendly design and robust features.

You can find detailed guides on connecting VitalPBX to MOR in the Kolmisoft Wiki.

8. Thirdlane

Thirdlane is an emerging PBX platform that is making waves by targeting many features traditionally found in 3CX. With a focus on multitenancy, Thirdlane offers a suite of solutions, including a multi-tenant PBX, unified communications, and contact center functionalities. It’s ideal for service providers looking for a modern alternative to 3CX with a range of enterprise-ready features.

9. Vodia PBX

Vodia PBX provides a multi-tenant platform similar to 3CX, but with a more complex licensing structure. Known for its cloud compatibility, Vodia offers extensive unified communications features that are suitable for service providers managing multiple clients. Its complex licensing can be a hurdle, but its robust capabilities make it worthwhile for larger providers.

10. PBXware

PBXware by Bicom Systems is an enterprise-grade PBX solution that supports multi-tenancy and is designed for service providers and large businesses. With versions for call centers, businesses, and multi-tenant environments, PBXware is a versatile platform offering the stability and scalability needed for professional deployments.

11. Wildix

Wildix is a web-based PBX that focuses on unified communications and offers strong collaboration tools, including video conferencing, chat, and integration with third-party software. It’s designed to work seamlessly with various devices and offers an intuitive user experience, making it popular among businesses with strong remote work requirements.

12. Wazo

Wazo is an open-source PBX platform built on Asterisk, offering an API-first approach to building unified communications systems. Wazo’s modular architecture and robust API support make it an excellent choice for developers and service providers who want to build tailored communication solutions.

13. Yeastar

Yeastar offers a range of hardware and software-based PBX solutions, providing flexibility in deployment. Known for their user-friendly interfaces and straightforward setup, Yeastar PBXs are suitable for small to medium businesses, offering unified communications, call center features, and CRM integrations.

14. PortSIP

PortSIP is a software PBX platform that emphasizes mobile and WebRTC support, catering to modern VoIP environments where mobility and video communication are essential. PortSIP’s focus on video, voice, and chat integration makes it ideal for service providers looking to offer a complete UC experience.

15. Grandstream UCM

Grandstream UCM is a hardware-only PBX that provides a robust, enterprise-grade communication solution. With its straightforward deployment and user-friendly management interface, Grandstream UCM is ideal for businesses seeking reliable, appliance-based PBX solutions.

16. Issabel

Issabel is a continuation of the Elastix PBX project, providing a feature-rich, open-source PBX platform built on Asterisk. It includes contact center features, CRM integration, and reporting tools, making it ideal for small to medium businesses that need an all-in-one communication solution.

17. Netsapiens (Crexendo)

Netsapiens is a carrier-grade, multi-tenant PBX solution designed for service providers requiring a scalable, high-availability platform. With its robust multi-tenancy and customizability, Netsapiens caters to large VoIP deployments, making it a top choice for providers handling multiple clients across geographies.

18. iVozProvider

iVozProvider is a multi-tenant PBX platform built with Kamailio and Asterisk, combining several open-source tools for a flexible, scalable VoIP solution. Its multi-tenancy features make it an good choice for service providers, particularly those familiar with Kamailio’s advanced routing capabilities.

Choosing the Right PBX Platform

Each of these PBX platforms offers unique advantages, and the right choice depends on your business’s needs, such as scalability, multi-tenancy, support requirements, and budget constraints. For instance:

• Small to Medium Businesses may prefer platforms like FreePBX, VitalPBX, or Issabel for their simplicity and community support.
• Large Enterprises and Service Providers might opt for multi-tenant solutions such as 2600hz Kazoo, or Netsapiens to manage multiple clients effectively.
• Unified Communications Focused Providers can consider 3CX, Wildix, or PortSIP for comprehensive UC solutions, including video, chat, and mobile support.

Whether you need open-source flexibility, commercial-grade support, or advanced multi-tenancy, this list offers a robust starting point to find a PBX platform that meets your VoIP service goals.

Explore these platforms, consider their unique features, and leverage the right tools to provide an outstanding VoIP experience for your customers.

If you need a class 4/5 softswitch, SBC or a billing platform for any of these PBXes, you can check out MOR and M4 softswitches:

• MOR features and description
• M4 features and description
• MOR vs M4 Comparison Table
• M4 Online Demo
• MOR Online Demo

The post Top PBX Platforms for VoIP Service Providers appeared first on Kolmisoft Blog.

Time-based routing is critical for VoIP providers, especially those working with call centers that rely on dialer calls. Strict regulations often limit dialer call hours, making manual routing not only time-consuming but also risky due to potential errors and compliance issues.

With MOR Softswitch and its Time Period-Based Least Cost Routing (LCR) feature, you can automate routing for dialer calls, ensuring compliance, reliability, and operational efficiency. In this blog, we’ll show you how MOR simplifies time-based routing for VoIP providers managing these challenges.

The Challenge: Managing Time-Based Routing Manually

Time-based routing is a critical component of VoIP operations, allowing service providers to comply with country-specific regulations, manage client needs, and optimize call flows. However, manual management of time-based routing is often inefficient and error-prone.

Consider these common scenarios:

• Manual Calls: These are routed through designated trunks without time restrictions.
• Dialer Calls: These must adhere to regulated time periods (e.g., 10:00 AM to 1:00 PM and 2:00 PM to 8:00 PM, Monday through Friday) and be blocked outside these hours to avoid penalties.

When routing is managed manually, it consumes valuable time and increases the risk of errors. Even a small mistake in configuration can lead to unauthorized calls, regulatory fines, and a loss of client trust.

Why Automating Time-Based Routing is Essential

Manual routing methods are no longer sustainable for modern VoIP businesses due to:

1. Strict Regulatory Compliance: Many countries impose regulations on when dialer calls can be made, making accurate routing essential.
2. Operational Inefficiency: Daily manual adjustments to trunks take up valuable resources and time.
3. Risk of Human Error: Mistakes in manual routing can lead to calls being placed outside allowed hours, resulting in regulatory breaches and fines.

The solution lies in automating these processes with tools designed for precision, flexibility, and compliance.

Introducing MOR Softswitch for Automated Time-Based Routing

The MOR Softswitch simplifies time-based routing through its Least Cost Routing (LCR) with Time Periods functionality. This powerful feature enables VoIP providers to define and automate time-specific routing for calls, ensuring compliance and improving operational efficiency.

Key Features of MOR’s Time Period-Based LCR

1. Customizable Time Periods: Define up to five unique time periods per LCR to meet specific client needs or comply with regulations.
2. Server Timezone Integration: Configurations are synchronized with the server’s Linux timezone for consistent performance.
3. Seamless Fallback to Main LCR: When calls fall outside defined time periods, they automatically route through the main LCR, ensuring uninterrupted service.

By using MOR Softswitch, providers can eliminate manual adjustments and maintain a seamless, error-free call routing experience.

How to Use MOR Softswitch for Time-Based Routing

Example 1: Complying with Dialer Call Regulations

Set up routing for dialer calls with specific time periods:

• Weekdays (Monday to Friday):
  • 10:00 AM to 1:00 PM
  • 2:00 PM to 8:00 PM

During these hours, the designated trunk for dialer calls is active. Outside of these periods, calls are automatically rejected, ensuring compliance.

Example 2: Seasonal Campaign Call Routing

For temporary marketing campaigns, you can set routing parameters like:

• Date Range: January 1st to February 28th
• Time Range: 9:00 AM to 6:59 PM

This configuration directs campaign-related traffic through specific trunks, enhancing efficiency during the promotional period.

Example 3: Managing After-Hours Calls

To ensure 24/7 availability, configure after-hours routing:

• Time Range: 7:00 PM to 8:59 AM (daily)

Calls outside standard business hours are seamlessly routed through designated trunks without manual intervention.

Benefits of Using MOR Softswitch

1. Enhanced Regulatory Compliance: Automating call routing ensures adherence to restricted calling hours, protecting providers from fines and penalties.
2. Improved Operational Efficiency: Save time and resources by automating repetitive tasks like manual trunk configuration.
3. Flexibility for Any Use Case: Easily set up routing schedules for regulatory compliance, marketing campaigns, or after-hours service.
4. Increased Reliability: Maintain uninterrupted service with automatic transitions between time periods and main LCR settings.

Why MOR Softswitch is the Best Choice for VoIP Providers

If you’re a VoIP provider seeking to optimize operations, reduce errors, and ensure compliance, MOR Softswitch offers a reliable and feature-rich solution. Its Time Period-Based LCR feature simplifies complex routing needs, making it ideal for businesses of all sizes.

By leveraging MOR Softswitch, you can:

• Eliminate manual routing processes.
• Ensure compliance with international call regulations.
• Enhance operational efficiency and reduce costs.

Conclusion: Automating Time-Based Routing for Success

Time-based routing no longer has to be a manual, error-prone process. With MOR Softswitch, VoIP providers can automate routing, ensure compliance, and improve client satisfaction. Whether you’re managing regulated dialer calls, seasonal campaigns, or after-hours traffic, MOR’s Time Period-Based LCR is the right solution for your operations.

Ready to improve your VoIP services? Visit Kolmisoft to learn more about how MOR Softswitch can help with your time-based routing and take your business to the next level.

• Time Period-Based LCR documentation
• MOR Features and Description
• MOR Online Demo

The post Automating Time-Based Routing for Dialer Calls with MOR Softswitch appeared first on Kolmisoft Blog.

In the fast-evolving world of telecommunications, businesses are continuously searching for more efficient, cost-effective, and secure ways to authenticate users. One such innovation gaining traction is flash call authentication—a method that leverages missed calls for verification instead of traditional SMS OTPs. This shift presents a significant opportunity for VoIP service providers to tap into a growing market while offering businesses an alternative to expensive messaging services.

With increasing digital transactions, rising concerns over SMS fraud, and enterprises looking to reduce operational costs, flash calls are becoming a preferred authentication method. In this article, we’ll explore how flash calls work, why they are gaining popularity, and the key opportunities they offer for VoIP service providers looking to expand their business.

What Are Flash Calls?

Flash calls are a growing trend in authentication services, where an automated call is used to verify a user’s identity instead of traditional SMS one-time passwords (OTPs). In this process, a user initiates authentication, and instead of receiving a text message, they receive a missed call from a designated number. The last few digits of this number serve as the verification code.

This method is increasingly being adopted by companies looking for more cost-effective, secure, and user-friendly authentication solutions. As a result, flash calls are becoming an essential part of the VoIP industry, creating new revenue streams and business opportunities for VoIP service providers.

Why Are Flash Calls Gaining Popularity?

1. Cost-Effectiveness

Compared to SMS OTPs, flash calls are significantly cheaper. By adopting flash call authentication instead of relying on costly SMS termination, businesses can streamline their verification processes while cutting operational costs.

2. Security & Fraud Prevention

Flash calls reduce the risk of SIM swap fraud and phishing attacks. Unlike SMS OTPs, which can be intercepted or spoofed, flash call authentication links the verification process directly to the user’s phone number.

3. Improved User Experience

Flash calls are faster and more seamless than SMS-based authentication. Users don’t need to manually enter an OTP; they just need to read the last digits of an incoming call’s number.

4. Growth in Mobile Authentication Market

With increasing digital transactions, banking, e-commerce, and other industries requiring secure authentication, the demand for flash call verification services is on the rise.

Opportunities for VoIP Service Providers

1. Monetizing Flash Call Termination

VoIP providers can route and terminate flash calls for enterprises and authentication platforms. This presents a revenue opportunity, especially for providers with strong interconnect agreements and access to direct routes.

2. Partnering with Enterprises & OTT Platforms

Companies offering authentication services—such as fintech firms, ride-hailing apps, and social media platforms—are looking for cost-effective and reliable flash call solutions. VoIP providers can establish direct partnerships to offer high-quality termination services.

3. Offering Flash Call Solutions as a Service

VoIP businesses can create APIs and software solutions that integrate flash call authentication into enterprise applications, similar to how Twilio and Vonage offer SMS authentication APIs.

4. Expanding into Emerging Markets

Many developing countries experience high SMS fraud rates and expensive messaging costs. VoIP providers can capitalize on these markets by providing flash call authentication as an alternative to SMS OTPs.

5. Enhancing Carrier Relations and A2P Traffic Handling

As flash calls generate more traffic between VoIP carriers and telecom operators, service providers can leverage this trend to improve business relationships, negotiate better termination rates, and increase overall A2P (Application-to-Person) traffic volume.

6. Providing Hybrid Authentication Services

Flash calls can be bundled with traditional VoIP services such as SIP trunking, number hosting, and enterprise telephony, creating a comprehensive authentication-as-a-service solution for businesses.

Challenges to Consider

• Regulatory Restrictions: Some telecom regulators may impose restrictions on flash calls, similar to A2P SMS regulations.
• Number Blocking by Operators: Mobile carriers may block flash call numbers to protect their A2P SMS revenues.
• Network Latency & Call Completion Issues: Ensuring high call completion rates and quick call setup times is critical for maintaining service quality.

Conclusion

Flash calls are transforming the way businesses authenticate users, and VoIP service providers have a unique opportunity to capitalize on this growing trend. By offering flash call termination, partnering with enterprises, and developing innovative authentication solutions, VoIP businesses can unlock new revenue streams and strengthen their market position.

As the demand for secure and cost-effective authentication continues to grow, now is the perfect time for VoIP providers to explore flash call opportunities and establish themselves as key players in this evolving space.

Kolmisoft does not currently offer billing for flash calls. However, we are closely monitoring market trends and client needs. If demand for this functionality increases, we will consider implementing it in future updates. Kolmisoft clients with a Support System login can cast their vote for the Flash Call billing functionality in the following development request: [Flash Calls] Bill all call attempts on certain trunks, even failed calls (#573).

The post Flash Calls in the VoIP Business: Opportunities for VoIP Service Providers appeared first on Kolmisoft Blog.

As the telecommunications industry evolves globally, we recently had an enlightening conversation with a client operating the international voice gateway in a West African nation. Their situation provides valuable insights for anyone navigating regulated telecom markets or dealing with declining traditional voice traffic.

The Changing Landscape of Voice Traffic

“Voice call or voice business is slowly degrading,” our client explained during our call. “Both incoming and outgoing. Because of technology, people have free options – social media, WhatsApp with video free.”

This is a reality facing telecom operators worldwide. However, in developing markets, there’s still significant value in traditional voice services. As our client noted, “We are fortunate that we are in a less developed country where they still have traditional mobile units. Not most of them have smartphones to communicate.”

This creates an interesting window of opportunity – traditional telecom services remain essential while digital transformation gradually progresses.

Operating in Highly Regulated Markets

One fascinating aspect of this business is navigating government regulations. In many developing countries, telecom is heavily regulated, sometimes with a single authorized international gateway.

“We cannot decide unless they approve it,” the client mentioned. “We are fully dependent on their approval all the time.”

This regulatory framework creates both challenges and opportunities:

Challenges:

• Limited pricing flexibility
• Restrictions on service diversification
• Approval processes for technical changes

Opportunities:

• Protected market position
• Stable business environment
• Governmental support for essential infrastructure

The Infrastructure Challenge

Perhaps the most interesting insight came when discussing local infrastructure reliability:

“There’s only one internet provider here in the country. So, in case this guy has a problem, everybody has a problem. You don’t have option B.”

This reality forces businesses to maintain hybrid solutions – keeping critical infrastructure both locally and internationally as backup. While this increases costs, it ensures business continuity in markets with developing infrastructure.

Diversification Strategies

When traditional voice traffic declines, where do you go? Our client is exploring several paths:

1. Regional interconnection: “We’re trying to interconnect with neighboring African regional countries directly to local MNOs.”
2. SMS integration: While currently, SMS traffic bypasses their gateway, this represents a potential growth area.
3. New technologies: Solutions like flash calls (using call initiation for authentication instead of SMS) could offer new revenue streams.

The key learning here is to stay adaptable while working within regulatory constraints.

Lessons for Telecom Businesses

Whether you’re operating in a developing market or an established telecom environment, several principles emerge:

1. Embrace hybrid infrastructure: Maintaining redundancy across different geographic locations ensures business continuity.
2. Work within regulatory frameworks: Understanding governmental priorities helps navigate restrictions successfully.
3. Pursue strategic diversification: Identify complementary services that leverage your existing infrastructure and relationships.
4. Value long-term partnerships: Our client has maintained the same technology partnership for nearly 8 years, creating stability in a changing market.

The telecommunications landscape will continue evolving, but the fundamentals of reliable service, strategic partnerships, and adaptable business models remain constant – whether you’re operating in New York or a small regulated market in West Africa.

The post Voice Gateway Business in Developing Markets: Challenges and Opportunities appeared first on Kolmisoft Blog.

In this video guide, we walk you through the process of integrating the 3CX phone system with the MOR Softswitch to enable seamless outgoing calls and real-time billing. Whether you’re setting this up for the first time or looking to verify your configuration, this tutorial covers all the essential steps — from SIP trunk setup to call verification and CDR inspection.

What’s Covered in the Video?

System Prerequisites

Before jumping into configuration, make sure you have:

• 3CX server installed and configured

• 3CX desktop application ready

• A user extension (e.g., 1000) active

• A fully configured MOR user and device with pre-set billing parameters

Step-by-Step Process

Step 1: Launch 3CX

Start by opening the 3CX interface so you’re ready to make test calls after setup.

Step 2: Configure SIP Trunk to MOR

• Go to the “Voice & Chat” section in the 3CX management console

• Click Add Trunk → select Generic VoIP Provider

• Enter:

  • A name for the trunk

  • MOR server address

  • SIP credentials (username/password)

• Configure the registration parameters as shown in the video

• Save the trunk configuration

Step 3: Create an Outbound Rule

Set up an outbound call rule in 3CX to route external calls through the MOR trunk.

Step 4: Verify Registration

• In 3CX, ensure the trunk is shown as registered (note: a warning/exclamation icon may appear since MOR is not on the 3CX tested list — this is expected).

• In MOR, navigate to the Devices section to confirm the 3CX connection appears as registered.

Step 5: Make a Test Call

Using the 3CX app, dial an external test number. If configured correctly, the call will:

• Connect successfully

• Appear in the Active Calls section in both 3CX and MOR

Step 6: Verify CDR and Billing

After completing the call:

• In MOR, check the CDR (Call Detail Records) to confirm billing occurred as expected.

• In 3CX, check the Call History to verify timestamps, durations, and call routing.

Why Integrate 3CX with MOR?

This setup allows you to:

• Use 3CX’s user-friendly interface and management tools

• Leverage MOR’s robust billing engine for prepaid, postpaid, or wholesale scenarios

• Monitor active calls and usage across both systems in real time

Final Thoughts

Connecting 3CX and MOR softswitch is a powerful way to streamline voice traffic routing and monetize outbound calling. With the right configuration, you can achieve a seamless experience for users and administrators alike.

Need help setting this up? Reach out to our support team — we’re happy to assist.

The post How to Connect 3CX with MOR Softswitch for Outgoing Calls and Billing appeared first on Kolmisoft Blog.

In today’s interconnected VoIP environment, being able to seamlessly integrate different systems is crucial for effective telecommunications management. One common requirement is forwarding Direct Inward Dialing (DID) calls from your billing system to your PBX. In this guide, we’ll walk through the process of setting up call forwarding from the MOR billing system to a 3CX phone system.

Why Integrate MOR with 3CX?

Before diving into the technical steps, let’s understand why this integration is valuable:

• Centralized Billing: All calls pass through MOR first, ensuring proper billing and reporting
• Enhanced Call Tracking: Maintain visibility of calls across both platforms
• Flexible Routing: Configure advanced routing rules in MOR while leveraging 3CX’s PBX capabilities
• Seamless User Experience: End users benefit from 3CX’s feature-rich environment

Step-by-Step Configuration Process

We’ve created a comprehensive video tutorial (embedded below) that demonstrates the complete process. Here’s a breakdown of the key steps:

1. MOR Configuration

First, we need to prepare the MOR billing system:

1. Add the DID Number: Enter the DID number provided by your telecom carrier into MOR
2. Reserve the DID: Assign this DID to a specific 3CX user
3. Configure the Trunk: Connect the DID to the appropriate 3CX trunk

Once these steps are completed, MOR is ready to forward incoming calls to your 3CX system.

2. 3CX Configuration

Now, let’s configure the 3CX system to receive these calls:

1. Add the DID to 3CX: Configure the same DID number in the 3CX system
2. Assign to User: Connect the DID to the desired user account
3. Verify Configuration: Ensure the DID appears under user settings
4. Set Up Call Routing: Optionally assign the DID to specific extensions or call routes

3. Testing the Integration

The final step is to verify everything works correctly:

1. Place a Test Call: Call the DID number from an external line
2. Verify in MOR: Confirm the call appears in MOR’s active calls section
3. Verify in 3CX: Check that the call properly reaches the 3CX system

Video Demonstration

Watch our detailed video guide for a visual walkthrough of the entire process:

Benefits of This Integration

By forwarding DID calls from MOR to 3CX, you gain several advantages:

• Complete Call History: All calls are logged in both systems
• Accurate Billing: MOR handles call accounting while 3CX manages the call experience
• System Redundancy: Two separate systems track the call flow
• Enhanced Reporting: Generate comprehensive reports from both platforms

Common Troubleshooting Tips

If you encounter issues with your integration, check these common problem areas:

• Trunk Registration: Ensure the 3CX trunk is properly registered in MOR
• Matching DIDs: Verify the DID number is identical in both systems
• Firewall Settings: Check that necessary ports are open between systems
• Codec Compatibility: Confirm both systems use compatible audio codecs

Conclusion

Integrating MOR with 3CX provides a powerful combination of billing capability and PBX functionality. This setup ensures transparent tracking, accurate reporting, and seamless integration between your core telecom systems.

By following the steps outlined in our video and this guide, you can create a robust call-forwarding system that leverages the strengths of both platforms.

Have you implemented this integration in your environment? What challenges did you face? Share your experience in the comments below!

Looking for more advanced telecom integration solutions? Contact our team of experts for personalized assistance with your specific requirements.

The post How to Forward DID Calls from MOR to 3CX: Complete Setup Guide appeared first on Kolmisoft Blog.

VoIP termination is always evolving — and Startel International might have just cracked the next big thing. After years in the VoIP business (including wholesale), they’ve found a smarter, leaner path forward: VoIP termination using regular Android smartphones.

No SIM boxes. No bulky GSM gateways. Just smart software, cheap Androids, and a plug-and-play model that scales.

And it’s already working — live in multiple countries, with real revenue.

Why VoIP Termination Still Matters — If You Do It Right

Over the past decade, wholesale voice traffic has grown significantly, powered largely by VoIP termination. But the wholesale model comes with serious downsides: razor-thin profit margins, unreliable partners, and high payment risks.

Worse still, the market is full of shady practices — from Fake Answer Supervision (FAS) to route manipulation — making it hard to maintain quality or trust. It’s a volume-driven game riddled with complexity and bad actors.

In contrast, VoIP-GSM termination offers better margins, and today’s mobile technology advancements enable simpler, stealthier, and far more reliable termination networks.

How the Smarter VoIP Termination System Works

At the heart of this system is a simple idea: turn regular Android phones into fully functional VoIP-GSM gateways — with no need for SIM boxes, gateways, or static installations.

After years of iteration, Startel International developed a custom Android app that transforms a smartphone into a VoIP-GSM gateway. The app connects directly to the MOR Softswitch using SIP credentials (not IP authentication), which makes the setup highly dynamic and easy to deploy over mobile data or Wi-Fi.

What the App Does

The Android app turns a standard phone into a smart, automated call handler — dialing GSM calls like a real human user, without needing any manual input. It uses the phone’s native dialer and SIM card, so there are no clunky audio bridges or unstable API workarounds. That’s why call quality is high and failure rates are low.

The app is engineered to:

• Simulate human behavior (random delays, varied durations, daily usage caps)
• Handle SIM popups, reboots, and call errors automatically
• Adapt to Android updates and device-specific quirks
• Run stealthily with behavioral randomization

The system is under constant development, evolving as telcos get smarter and environments change. Updates include stealth enhancements, new compatibility layers, and fine-tuned device logic to ensure long-term functionality.

How the Full Setup Works

Here’s a simplified look at how the full termination flow operates:

1. Phone Registration via SIP:
   Each Android device registers with the MOR Softswitch using SIP credentials. No fixed IPs required — ideal for roaming or mobile setups.
2. VoIP Call Routing via MOR:
   When a VoIP call comes in, MOR routes it to the phone like a regular SIP call. Each phone is treated as a separate provider within the MOR softswitch.
3. Physical GSM Call Execution:
   The app receives the SIP call and physically dials out using the SIM card — as if a real user tapped “Call.” No tricks, no relays — just native GSM calling.
4. Optional VPN (Country-Specific):
   VPNs like Voxility can be used in tougher regions, but in many cases, they aren’t needed at all.
5. Traffic Logic on MOR Side:
   All call logic — routing, prioritization, balancing — is handled within MOR. The phones simply receive calls and pass them through the GSM network.
   

Why This System Beats Traditional Termination

Traditional VoIP-GSM setups haven’t changed much in over a decade. They rely on bulky SIM boxes and gateways that are easy to detect, expensive to maintain, and hard to scale. These systems are rigid, vulnerable, and increasingly ineffective in high-risk environments.

In contrast, the Android-based system developed by Startel International flips the script. It offers a smarter, more resilient model — one that prioritizes stealth, simplicity, and flexibility without sacrificing control or quality.

Here’s why it works so well — and why it’s better than traditional termination:

Harder to Detect
Phones behave like normal users. They move, change networks, and don’t trigger telco alarms like fixed gateways do. There’s no static location or base station to track.

True Mobility
Android devices can be physically relocated — even across borders — breaking geo-fencing and static profiling. Roaming SIMs? No problem.

Human-like Behavior
Randomized call patterns, varied durations, SIM rotation, and daily limits simulate real users — extending SIM life and reducing detection.

No SIM Fingerprinting
Because the system uses regular Android phones (not custom hardware), telcos can’t easily fingerprint them or distinguish them from consumer devices.

Clean, Trusted Traffic
No spy calls. No junk routes. Only whitelisted, stable VoIP traffic flows through the system — reducing SIM exposure and improving profitability.

No More IP Auth Headaches
Phones connect via SIP credentials, not IP-based authentication — ideal for mobile networks and CG-NAT environments.

Scalable and Lightweight
No racks or routers needed. A backpack can carry 10+ phones. Scaling is as simple as plugging in another Android.

Quick Recovery
If a phone is blocked, just replace it. Install the app, plug in a new SIM, register via MOR Softswitch, and you’re back online in 15 minutes.

Designed for Tough Markets
This setup thrives where traditional gear fails — from Africa to South Asia. It’s built to survive in hostile telecom environments.

Want to Join? Here’s the Model

Local partners set up the phones. Startel handles everything else.

Who’s It For?

• VoIP players already working with SIMs or GSM gateways
• Operators with local SIM access and decent power/connectivity
• Anyone tired of unreliable upstreams, static gear, or unreliable traffic

You (Local Partner) Provide:

• Android phones (used or cheap models work)
• Local SIMs
• Basic setup and monitoring (battery, signal, uptime)
• Occasional device movement to reduce detection
  

Startel Provides:

• Custom Android termination app
• MOR-based traffic routing
• Assistance for getting stable VoIP traffic (no spy calls)
• Ongoing app updates and tech support
• Transparent revenue-sharing model

Device Compatibility

The current setup has been successfully tested on several Android devices: Samsung A53, A54, A15, and A16 running Android 13 or 14. These phones are already in production and working well. Device testing is ongoing, and new models are added as they are validated in real-world deployments.

SIM Top-Up & Management

The Local Partners currently handle SIM recharges independently, usually via local cash top-ups, which gives them flexibility and control. It is also possible to open an account with an international recharge platform that allows sub-account access—ideal for resellers or partners who want a unified recharge system.

There are also plans to automate SIM warehouse tracking (to monitor stock, rotations, and recharge status) and top-up scheduling, so partners can manage SIM lifecycles and replenishment with minimal manual work.

Real Deployments — Already Live:

This isn’t just theory. The system is live in real-world conditions:

• Italy – 20+ phones online, profitable
• Ethiopia – Tough market, stealth setup holds
• Yemen – High-value traffic, still undetected
• Pakistan – Frequent SIM movement keeps traffic flowing
• Ivory Coast – Local partner setup running well
• Croatia (Roaming) – Croatian SIMs terminating from Italy with great results

Next up: Nigeria, Algeria, Tunisia, France — and more.

What’s Next?

Startel’s goal is ambitious: 1,000+ Android phones live by year-end.

With every new deployment, the software evolves. Bugs are fixed fast, features are added, and performance improves daily.

The network is growing, and so are partner earnings. This isn’t a prototype — it’s a proven business model with active, profitable operations.

Ready to Launch? Contact Startel to become a local partner.

Want to learn more about MOR Softswitch? Contact Kolmisoft.

The post Android-Powered VoIP Termination — No SIM Boxes. No Headaches appeared first on Kolmisoft Blog.

In this tutorial, we’ll demonstrate how to configure an outgoing call from a MicroSIP softphone through VitalPBX, using MOR Softswitch for billing and call management.

Why Integrate VitalPBX with MOR Softswitch?

Combining VitalPBX with MOR Softswitch allows for efficient call routing and comprehensive billing capabilities. This integration ensures that outgoing calls are managed effectively, with detailed billing records maintained for analysis and accountability.

Step-by-Step Configuration Guide

1. Prerequisites Setup

Ensure the following components are created in MOR:

• User: vitalpbx123
• Device: vitalpbx

2. Configure VitalPBX Extension

1. Navigate to Extensions in VitalPBX.
2. Select an available extension (e.g., 1001).
3. Note the password for this extension for later use.

3. Configure MicroSIP Softphone

1. Open MicroSIP and click Add Account.
2. Enter the following settings:
   • Account Name: Descriptive name
   • SIP Server: VitalPBX IP address
   • Username: 1001
   • Domain: Same as SIP Server
   • Login: 1001
   • Password: Password from VitalPBX extension
   • Display Name: Preferred display name
3. Click Save.

MicroSIP should now display as online, indicating successful registration.

4. Create SIP Trunk to MOR Softswitch

1. In VitalPBX, navigate to External → Trunks.
2. Create a new trunk with the following settings:
   • Description: MOR Trunk
   • Remote Username: Username from MOR configuration
   • Remote Secret: Password from MOR configuration
   • Remote Host: MOR Softswitch IP address
   • Local Username: Same as Remote Username
   • Remote Port: 5060
   • Identify By: Auth Username and IP
3. Click Save and Apply Config.

Verify in MOR that VitalPBX has successfully registered.

5. Configure Outbound Routes

1. Navigate to External → Outbound Routes.
2. Configure the route with the following settings:
   • Description: Descriptive name for the route
   • Trunk: Select the newly created MOR Trunk
   • Pattern: X. (matches all outgoing calls)
3. Click Save and Apply Config.

6. Test the Call Flow

Using MicroSIP, dial an external number to test the configuration. The call should connect successfully, indicating that the setup is functioning correctly.

7. Monitor and Verify

Monitor the call in real-time through:

• MOR Active Calls: View the call in MOR’s Active Calls window.
• VitalPBX Monitoring: Check the active calls in the VitalPBX interface.

8. Review Billing and Call Details

After the call completes:

1. In MOR, review how the call was billed.
2. Access detailed call information, including:
   • Call duration
   • Billing rates applied
   • PCAP logs for troubleshooting

Conclusion

By following this guide, you’ve successfully configured a complete call flow from MicroSIP softphone through VitalPBX to MOR Softswitch, with proper billing and monitoring capabilities. This setup provides a robust foundation for VoIP services with comprehensive call management and billing features.

Video Demonstration

Watch our detailed video guide for a visual walkthrough of the entire process:

Need a more advanced telecom integration?

Our experts are here to help you design the perfect setup tailored to your business needs. Contact us for a personalized consultation.

Want to dive deeper? Check out these helpful guides:

• Top PBX Platforms for VoIP Service Providers – Discover the best options for your VoIP business.
• Forwarding DID Calls from MOR to 3CX – Seamlessly handle inbound calls with MOR and 3CX.
• Connecting 3CX with MOR for Outgoing Calls and Billing – Simplify your outbound call flow and ensure proper billing.
• 3CX Removed Us from the Supported SIP Trunk Providers Page – What to Do? – Understand the implications and your next steps.
• PBX Security Best Practices – Protect your infrastructure from common VoIP threats.

The post Billing Outgoing Calls from VitalPBX through MOR Softswitch appeared first on Kolmisoft Blog.

Современные VoIP-операторы и IT-компании все чаще выбирают гибкие и мощные решения для телефонии и биллинга. Комбинация биллинговой платформы MOR от Kolmisoft и популярной АТС 3CX дает возможность достичь именно этого — надежной тарификации и комфортного управления звонками.

В этом материале, основанном на двух обучающих видео, мы рассмотрим, как настроить двустороннюю маршрутизацию вызовов между MOR и 3CX.

Входящая маршрутизация: Направление DID-звонков из MOR в 3CX

Процесс настройки входящих звонков из MOR в 3CX состоит из следующих шагов:

1. Добавление DID в MOR

Вы начинаете с DID-номера, полученного от оператора. Этот номер добавляется в биллинговую систему MOR и резервируется за нужным пользователем, к которому будет направлен звонок.

2. Назначение DID на SIP-транк 3CX

После резервирования вы назначаете DID на соответствующий SIP-транк, связанный с 3CX. Таким образом, MOR готов перенаправить звонки.

3. Настройка 3CX для приема звонков

В интерфейсе 3CX вы добавляете тот же DID-номер к нужному пользователю или задаёте правила маршрутизации для направления звонков на нужный внутренний номер.

4. Тестовый звонок и проверка

Совершив тестовый звонок, вы убедитесь, что вызов успешно проходит, отображается в обеих системах (MOR и 3CX), и подлежит корректной тарификации.

Такой подход позволяет вам:

• централизованно тарифицировать входящие звонки;
• сохранять прозрачность и учет;
• легко управлять маршрутизацией и пользователями.

Исходящая маршрутизация: Направление звонков из 3CX в MOR

Теперь рассмотрим, как настроить исходящие вызовы из 3CX через биллинговую платформу MOR:

1. Подготовка систем

3CX уже установлена и настроена на сервере. В MOR настроены SIP-аккаунт и параметры тарификации.

2. Создание SIP-транка в 3CX

В консоли управления 3CX:

• Добавьте новый VoIP-провайдер (Generic).
• Укажите IP-адрес MOR и данные аутентификации.
• Настройте регистрацию и сохраните изменения.

3. Создание правила маршрутизации

Создайте правило, направляющее определенные номера или диапазоны на MOR-транк.

4. Проверка регистрации

Убедитесь, что транк зарегистрирован. В 3CX он может отображаться с предупреждением (т.к. MOR не является сертифицированным провайдером), но это нормально. На стороне MOR вы также увидите зарегистрированное устройство.

5. Совершение тестового звонка

Используйте приложение 3CX, чтобы позвонить на внешний номер. Убедитесь, что звонок активен и отображается в обеих системах.

6. Проверка CDR и истории вызовов

В MOR проверьте CDR (Call Detail Records), а в 3CX — журнал звонков. Сравните данные и убедитесь, что тарификация прошла корректно.

Заключение: Почему стоит интегрировать MOR и 3CX

Интеграция MOR и 3CX — это мощный инструмент для тех, кто хочет получить:

• Современную АТС с удобным интерфейсом
• Гибкую маршрутизацию и контроль
• Надежную тарификацию входящих и исходящих вызовов
• Поддержку биллинга в режиме реального времени

Эта связка особенно полезна для VoIP-операторов, IT-компаний и бизнесов, работающих с SIP-телефонией.

Если у вас возникли вопросы по настройке или нужна помощь, обратитесь в нашу службу поддержки или изучите подробную документацию Kolmisoft.

Полезные материалы:

• Интеграция MOR и 3CX: входящие вызовы (видео)
• Интеграция MOR и 3CX: исходящие вызовы (видео)

Свяжитесь с нами, если вы хотите протестировать такую интеграцию или получить индивидуальную консультацию.

The post Интеграция MOR и 3CX: Полное руководство по маршрутизации входящих и исходящих вызовов appeared first on Kolmisoft Blog.

Современная телефония требует гибкой маршрутизации, прозрачной тарификации и надежной интеграции между системами. В этой статье мы пошагово покажем, как объединить возможности IP-АТС VitalPBX с биллинговой платформой MOR Softswitch, чтобы обеспечить полный контроль над VoIP-звонками — от исходящего вызова до приема входящего звонка на DID.

Часть 1: Исходящие вызовы из VitalPBX через MOR

1. Подготовка в MOR

Создайте пользователя vitalPBX123 и устройство с тем же именем. Отметьте его как транк (SIP trunk).

2. Настройка VitalPBX

В разделе Extensions создайте абонента (например, 1001) и сохраните его SIP-пароль.

3. Настройка MicroSIP

Подключите softphone к VitalPBX:

• SIP-сервер и домен: IP VitalPBX
• Имя пользователя и логин: 1001
• Пароль: из настройки абонента

4. Создание SIP-транка

В разделе External → Trunks настройте:

• Remote Secret: пароль из MOR
• Remote Username: логин MOR
• Remote Host: IP MOR
• Remote Port: 5060
• Идентификация: по Auth Username и IP

5. Настройка исходящего маршрута

В разделе Outbound Routes:

• Шаблон: . (все номера)
• Выберите транк MOR

6. Тестовый звонок

Совершите звонок через MicroSIP и проверьте:

• MOR: Active Calls и CDR
• VitalPBX: активные вызовы и Reports

Часть 2: Прием входящих звонков на VitalPBX через MOR

1. Назначение DID в MOR

В разделе DIDs:

• Зарезервируйте DID за нужным пользователем
• Назначьте на соответствующий транк (в сторону VitalPBX)

2. Настройка Inbound Route в VitalPBX

В разделе Inbound Routes настройте:

• Routing Method: Default
• Description и DID Pattern: номер DID
• Inbound Destination: Extensions → нужный номер (например, 1001)

3. Тестовый звонок

Позвоните на настроенный DID и проверьте:

• MOR: Last Calls
• VitalPBX: Reports → CDR

Вывод: Надежная связка для VoIP-услуг

Интеграция VitalPBX и MOR обеспечивает:

• Надежную маршрутизацию
• Прозрачную тарификацию
• Централизованный контроль
• Гибкую масштабируемость VoIP-решений

Для получения дополнительной помощи вы можете ознакомиться с нашей документацией.

 Полезные материалы:

• Исходящие звонки: настройка и тарификация
• Прием звонков через DID

Напишите нам, если хотите протестировать интеграцию или обсудить детали под вашу инфраструктуру.

The post Интеграция VitalPBX и MOR: Тарификация и маршрутизация входящих и исходящих вызовов appeared first on Kolmisoft Blog.