Boost Sales &
Agent Productivity
with the Ultimate
Hosted Dialer.

In the modern enterprise landscape Microsoft Teams has evolved into the primary communication hub for global organizations. While Teams provides a robust internal platform for voice and video its true power as a business phone system is unlocked when connected to the Public Switched Telephone Network (PSTN).

To bridge the gap between the Microsoft cloud and traditional telephony organizations must leverage SIP Trunking for Microsoft Teams. This guide explores how to implement this connectivity using Direct Routing, the benefits of Domestic Termination and the technical architecture required for a secure scalable deployment.

Understanding the Architecture: SIP Trunking & Direct Routing

At its core SIP (Session Initiation Protocol) is the signaling language of modern VoIP. A SIP Trunk is a virtual version of an analog phone line that allows a SIP provider to connect one or more channels to an organization’s PBX.

What is Direct Routing?

Direct Routing is the Microsoft endorsed mechanism that allows businesses to connect their own telecommunications provider (via a SIP Trunk) to the Microsoft Teams Phone System. Unlike Microsoft Calling Plans where Microsoft acts as the carrier Direct Routing allows you to “Bring Your Own Carrier” (BYOC).

The Role of the Session Border Controller (SBC)

In a Direct Routing setup the Session Border Controller (SBC) acts as the critical gatekeeper. It sits between the telephony provider’s SIP Trunk and the Microsoft Teams cloud. The SBC performs several vital functions:

• Security: Acts as a specialized firewall for voice traffic preventing toll fraud and DDoS attacks.
• Interoperability: Translates signaling and media between different protocols.
• Media Optimization: Ensures call quality by managing jitter latency and packet loss.

The Strategic Value of Domestic Termination

When selecting a SIP provider for Microsoft Teams the quality of Domestic Termination is often the deciding factor for enterprise grade reliability.

What is Domestic Termination?

Domestic Termination refers to the process of routing a call to its final destination within the same country where the call originated. For example if a Teams user in London calls a landline in Manchester a provider with strong UK Domestic Termination will route that call directly through local UK carrier interchanges.

Benefits of High Quality Domestic Termination:

• Superior Call Quality: By keeping the call path local you minimize the “hops” the data packets must take. This significantly reduces Latency and Jitter ensuring crystal clear audio.
• Drastic Cost Reduction: Local termination is almost always cheaper than international transit. Businesses can save up to 40 60% compared to standard global calling plans.
• Accurate Caller ID (CLI): Proper domestic routing ensures that local caller ID is presented correctly increasing call back rates for sales and support teams.
• Regulatory Compliance: Many regions (such as the EU and India) have strict data and telephony residency laws. Domestic termination ensures calls are handled within legal geographic boundaries.

Direct Routing vs. Operator Connect: Making the Choice

Microsoft offers three primary ways to achieve PSTN connectivity:

Verdict: Direct Routing remains the gold standard for enterprises that require complex call routing integration with legacy hardware (like elevator phones or pagers) or specialized Domestic Termination across multiple global regions.

Technical Requirements & Prerequisites

To successfully deploy SIP Trunking for Teams the following infrastructure must be in place:

Licensing

Users must have a Microsoft Teams Phone license. This is included in Microsoft 365 E5 or available as an add on for E1/E3 or Business Premium licenses.

Certified SBC

Microsoft only supports Certified Session Border Controllers. Leading vendors include:

• Ribbon Communications (SBC SWe Edge)
• AudioCodes (Mediant series)
• Cisco (CUBE)

Network and Security

• TLS 1.2: Microsoft enforces Transport Layer Security for all SIP signaling.
• Certificates: The SBC must have a certificate from a trusted Public Certificate Authority (CA). Self signed certificates will be rejected by the Microsoft SIP Proxy.
• Public IP & FQDN: The SBC requires a static public IP and a Fully Qualified Domain Name (FQDN) (e.g. sbc1.contoso.com). Note: *.onmicrosoft.com domains are not supported for SBC FQDNs.

Step by Step Configuration Guide

Configuring Direct Routing involves a mix of the Teams Admin Center (TAC) and PowerShell.

Step 1: Pair the SBC with the Tenant

Navigate to Voice > Direct Routing in the Teams Admin Center. Add a new SBC using its FQDN and specify the signaling port (usually 5061 for TLS).

Step 2: PowerShell Provisioning

Use the Teams PowerShell module to define the gateway.

PowerShell

New CsOnlinePSTNGateway  FQDN sbc1.contoso.com  SipSignalingPort 5061  MaxConcurrentCalls 100  Enabled $true

Step 3: Configure Voice Routing

You must create PSTN Usages Voice Routes and Voice Routing Policies.

• PSTN Usage: A label (e.g. “LocalCalls”) that links routes to policies.
• Voice Route: Uses Regex to determine which numbers go to which SBC.
  • Example Regex for US numbers: ^\+1(\d{10})$
• Voice Routing Policy: Assigned to users to grant them access to specific PSTN usages.

Optimization and Troubleshooting

Even with a perfect setup voice traffic is sensitive to network conditions.

Monitoring Call Quality (MOS Scores)

Monitor the Mean Opinion Score (MOS) via the Microsoft Call Quality Dashboard (CQD). A score of 4.0 or higher is considered “Excellent.” If scores drop check for:

• Bandwidth Throttling: Ensure voice traffic has priority via QoS (Quality of Service) tagging (DSCP 46).
• Firewall Blocks: Ensure ports 5061 (Signaling) and 10000–20000 (Media/RTP) are open.

Common SIP Errors

• 403 Forbidden: Usually indicates an FQDN mismatch or an unverified domain in the Microsoft tenant.
• One Way Audio: Typically a NAT (Network Address Translation) issue where the media ports are not correctly mapped on the firewall.

Conclusion

SIP Trunking for Microsoft Teams via Direct Routing is the most robust solution for modern enterprise communication. By selecting a provider that excels in Domestic Termination businesses can ensure crystal clear call quality while significantly reducing operational costs. As we move deeper into 2026 the integration of AI driven analytics within SBCs will further refine how we manage global voice traffic making Direct Routing an indispensable asset for the digital workplace

In the modern telecommunications landscape the difference between a high performing call center and a failing one is often measured in milliseconds. For Chris Maxwell an Operations Manager and Samantha Carter a Sales Lead the objective is simple: Maximum Human Connection.

However the reality of outbound dialing is that up to 70-80% of calls end up in voicemail. This is where Answering Machine Detection (AMD) becomes the most critical feature of a Hosted Dialer Solution. This guide provides an exhaustive look into the mechanics, the trade offs and the future of AMD technology.

What is Answering Machine Detection? 

Answering Machine Detection (AMD) is a sophisticated signal processing technology used by automated dialers to distinguish between a live human greeting and an automated recording.

The Evolution of AMD

Historically AMD relied on simple “energy detection.” If the sound lasted longer than a certain duration it was flagged as a machine. Today as part of the Hosted Dialer ecosystem AMD uses Neural Networks and Asynchronous Processing to analyze frequency cadence and even background noise to make a determination.

How AMD Works: The Technical Anatomy?

To understand the “How” we must look at the Audio Analysis phases. When a call is answered the Hosted Dialer begins a three stage observation:

Phase 1: The Initial Greeting 

• Human Cadence: Humans typically say “Hello?” or “Hello this is [Name].” This is usually short (under 2 seconds) and followed by a sharp silence as they wait for a response.
• Machine Cadence: Automated greetings are longer. “Hi you’ve reached the voicemail of… please leave a message after the beep.” This continuous stream of speech usually exceeds 2.2 seconds.

Phase 2: Silence Duration 

AMD monitors the silence after the initial greeting.

• The 700ms Rule: If the system detects a greeting followed by at least 700ms to 900ms of silence it classifies the call as a Live Person.
• The Timeout: If no silence is detected after 2.5 seconds of speech the system flags it as a Machine.

Phase 3: Frequency Recognition

Modern AMD can detect the “beep” of a voicemail or the “pre recorded” tone of an IVR system allowing for instant disconnection or a “Voicemail Drop.”

Technical Implementation: API Parameters for Developers

Using the Lexical Semantics found in top tier documentation (like Competitor 1) here is how you configure AMD for maximum Campaign Effectiveness.

Core Variables

• MachineDetection: The Boolean trigger. Setting this to Enable initiates the listening phase.
• AsyncAmd: Traditionally the dialer waits for AMD to finish before “bridging” the call to an agent. Asynchronous AMD bridges the call immediately but listens in the background cutting down the “dead air” time.
• MachineDetectionTimeout: Usually set between 3000ms and 5000ms. If the system cannot decide within this window it defaults to a specific action (usually “Human” to avoid losing a lead).

Advanced Tuning (The “Secret Sauce”)

• MachineDetectionSpeechThreshold: Determines how loud the audio must be to count as “speech.” This is vital for filtering out background white noise in a busy office.
• MachineDetectionSpeechEndThreshold: Defines how much silence is needed to conclude the greeting has ended.

The Accuracy vs. Speed Trade off

This is the “Pain Point” for Samantha Carter. If AMD is too aggressive it hangs up on real people (False Positive). If it is too slow the person says “Hello?” three times gets no answer and hangs up (Call Abandonment).

False Positives (The Profit Killer)

A False Positive occurs when a live human is identified as a machine. This happens if:

• The person has a long professional greeting (“Hello thank you for calling the law offices of Smith and Associates, how can I help you?”).
• There is significant background noise (dogs traffic) that the system mistakes for continuous speech.

False Negatives (The Productivity Killer)

A False Negative occurs when a machine is identified as a human. The agent hears: “…please leave your message after the tone.” This wastes 10-15 seconds of Agent Productivity.

Compliance and Call Abandonment Rates

In many jurisdictions (like the FTC in the US or Ofcom in the UK) Call Abandonment is a legal metric.

• If a human answers and there is no agent available within 2 seconds the call is considered “Abandoned.”
• Aggressive AMD can increase this delay. To stay compliant Hosted Dialers must balance the AMD analysis time with the “Time to Connect.”

Best Practices for Optimized Outreach

1. Use “Voicemail Drop”: Instead of just disconnecting, configure the dialer to play a pre recorded high quality message when a machine is detected.
2. Geographic Tuning: Greetings vary by culture. In some regions people answer with just “Yes?” (very short) while in others they are more verbose. Adjust your SpeechThreshold based on the target lead’s location.
3. Continuous Benchmarking: Use the Request Inspector to audit “Answered” vs “Detected” logs weekly.

FAQs

Can AMD distinguish between a residential answering machine and a corporate IVR?

Yes. Corporate IVRs often have a specific frequency and “Enter Extension” prompts that sophisticated AMD algorithms can identify through frequency pattern matching whereas residential machines follow a simple “Greeting  > Beep” structure.

How does 5G and VoIP latency affect AMD accuracy?

Jitter and latency in VoIP can “break” the audio stream. If the “Hello” is clipped the AMD might see it as two short bursts of sound instead of one leading to a False Positive. High quality Hosted Dialer Solutions use jitter buffers to stabilize audio before analysis.

Does “Background Noise Suppression” in the dialer help AMD?

Surprisingly no. If the suppression is too strong it can cut off the tail end of a human’s “Hello” making the speech seem shorter than it is. It is better to tune the SpeechThreshold within the AMD settings rather than relying on general noise cancellation.

Why do some agents hear the “Beep” even with AMD enabled?

This usually happens when the MachineDetectionTimeout is reached. If the machine’s greeting is exceptionally long (e.g. a 30 second instructional message) the system “gives up” and passes the call to the agent to be safe.

Is there a way to hide the “AMD Delay” from the customer?

Using Asynchronous AMD (AsyncAmd) is the best method. The agent is connected as the person is still saying “Hello.” The agent might hear the last syllable but the “dead air” is eliminated making the interaction feel natural.

In 2026 the telecommunications landscape has shifted. While proprietary “black box” dialers have become more expensive and restrictive, Open Source Dialer Software has evolved into an enterprise grade powerhouse. 

However for many businesses the word “free” is a paradox. While the software itself costs nothing to download the infrastructure maintenance and expert level configuration required to run it can create a massive hidden “Technical Debt.”

This article serves as a strategic roadmap for decision makers who are caught between the desire for total flexibility and the need for operational uptime.

Understanding VICIdial Asterisk and FreePBX

To navigate this topic one must understand the core entities that define the open source telephony ecosystem.

VICIdial: The Industry Standard

VICIdial is the most popular open source contact center suite in the world. It is built to interact with the Asterisk Open Source PBX to act as a complete inbound/outbound solution. It supports predictive dialing power dialing and manual dialing modes.

Asterisk: The Engine

Asterisk is the underlying framework. Think of it as the “engine” of the car while VICIdial is the “dashboard” and “controls.” Asterisk handles the SIP protocols call routing and media processing.

FreePBX: The GUI Specialist

While VICIdial is built for high volume call centers, FreePBX is often used for general office communications. For technical users integrating FreePBX with a dialer can provide a unified communication experience.

The “Chris Maxwell” Persona: Why Technical Skill Level Matters

We’ve identified our primary ICP as Chris Maxwell—a CTO or Lead DevOps Engineer who is “Technically Inclined.” Chris isn’t looking for a basic “how to” guide; he’s looking for a justification of Managed Services vs. Self Hosting.

Chris values:

• Root Access: He wants to know he can change the code if he needs to.
• API Extensibility: He needs the dialer to talk to his custom built CRM.
• Efficiency: He knows that his time is worth $150/hour so spending 10 hours fixing a SIP trunk issue is a $1500 loss for the company.

The Pros of Open Source Dialer Software

Unmatched Flexibility

Proprietary dialers (vs. Proprietary Dialer) often lock you into their ecosystem. With OSS if you need a specific dialing algorithm that prioritizes leads based on their “last clicked” timestamp in your CRM you can write a script to do it.

Scalability without Seat Licenses

The most significant attribute focus here is Cost. In a proprietary model adding 100 agents means adding $5000/month in seat fees. In an open source model you simply scale your cloud instances.

Community Driven Security

Because thousands of “Chris Maxwells” are looking at the code, bugs and security vulnerabilities are often patched faster than in proprietary systems.

The Cons of Self Hosted OSS

Complexity and TCO

The “Free” software requires a Linux Admin, a Database Optimizer and a VoIP Engineer. When you calculate the Total Cost of Ownership (TCO) the salaries of these experts often exceed the cost of managed hosting.

The Maintenance Burden

Self hosting means you are responsible for Security Updates. In 2026 SIP attacks are automated and relentless. One missed patch on your Asterisk server could lead to $20000 in fraudulent international calls in a single weekend.

Domestic Termination and Call Quality

One attribute often ignored by competitors like Nextiva is Domestic Termination. Domestic Termination is the ability to route calls to local numbers within your target country at high volume and low cost. When you self host you must find and vet your own SIP carriers. 

If your Domestic Termination provider has low quality your VICIdial predictive dialer will “over dial” causing high drop rates and burning through your leads. Managed providers often include “Carrier Grade” termination routes that are pre optimized for VICIdial’s aggressive dialing patterns.

Managed Rocky Dialer VICIdial Hosting: The Middle Path

This is where the Antonymy of “Open Source vs. Managed” becomes a “Hybrid Solution.” Managed VICIdial hosting (like the Rocky Dialer configuration) offers the best of both worlds.

• Expertise: You get the software for free but you pay for the hosting and management.
• Security: Proactive firewalls specifically tuned for VICIdial.
• Uptime: Redundant power and bandwidth that most mid sized businesses cannot afford on premise.

Scalability Metric: Self Hosted vs. Managed Hosting

To help Chris Maxwell make the final decision we provide this capacity analog.

Security Compliance and Information Responsiveness

In 2026 compliance with regulations like STIR/SHAKEN and TCPA is non-negotiable. An open source system must be meticulously configured to handle:

• DNC (Do Not Call) Scrubbing.

• Call Recording Laws.

• Shaken/Stir Identity Tokens.

Managed hosting provides Information Responsiveness by keeping these protocols updated automatically ensuring your business stays on the right side of the law.

Conclusion

For the technical user the “Problem” is the friction of maintenance. The “Solution” is Managed OSS. By choosing a managed environment for your VICIdial or Asterisk system you retain the flexibility that open source is famous for while gaining the stability of an enterprise cloud.

Explore Managed OSS dialer hosting and see when managed VICIdial infrastructure makes sense for your team.

FAQs

Can open source dialer software be customized without modifying core code?

Yes. Many open source dialers support customization through configuration layers  APIs  and external integrations. This approach allows teams to extend functionality while preserving upgrade compatibility.

How much technical maintenance does a VICIdial-based system typically require?

Maintenance requirements vary by scale  but production environments usually need ongoing monitoring  periodic updates  call quality tuning  and security reviews to remain stable over time.

Does open source dialer software support remote or distributed agents?

Open source dialers can support remote agents  but this typically requires additional network configuration  security controls  and bandwidth optimization to ensure call quality and system reliability.

What role does Asterisk play in open source dialer software?

Asterisk functions as the core telephony engine that handles SIP signaling  call routing  and media processing. Dialer platforms like VICIdial build on top of Asterisk rather than replacing it.

Are open source dialers suitable for regulated industries?

They can be  but compliance depends on how the system is configured and maintained. Logging  encryption  access controls  and update policies must be carefully implemented to meet regulatory requirements.

How long does it take to deploy an open source dialer system?

Deployment time can range from days to weeks depending on infrastructure readiness  customization needs  and technical expertise. Managed hosting significantly shortens this timeline.