Overview This guide outlines how to prepare Cisco routers and firewalls (ISR, ASA, Catalyst) for SIP-based VoIP. Cisco platforms are generally very reliable for VoIP, but SIP inspection and default firewall behaviour can cause issues if left unchanged.   Before you start You’ll need: CLI or web access (depending on platform) SIP provider details Bas...

Overview This guide covers how to configure a SonicWall firewall for SIP-based VoIP services. SonicWall works well with VoIP, but only if a few default features are adjusted. In most cases, issues come down to SIP transformations (ALG), NAT behaviour, or blocked RTP traffic. If these aren’t set correctly you’ll typically see: One-way audio Calls dro...

This article provides a step-by-step guide on how to review call logs and view call flows 1. Log in to the Connect Dashboard  ( https://customerportal.hp2k.co.uk/ )  Once logged in, select “Logs” from the left‑hand menu to open the Call Logs Overview page.   2. Filtering the Call Logs On the Call Logs Overview: Open the drop‑down filter and select I...

Overview This guide covers how to configure MikroTik routers for SIP-based VoIP. MikroTik is very flexible, but that flexibility comes with a catch — the defaults aren’t always VoIP-friendly. Things like SIP helper, FastTrack, and connection tracking can all interfere if left as-is. Once set up properly though, MikroTik is very stable.   Before you ...

Overview This guide covers WatchGuard firewalls for SIP-based VoIP. The main thing with WatchGuard is avoiding SIP proxy policies — packet filters are usually the better option.   Before you start Access to Firebox (Web UI or Policy Manager) SIP provider details   Ports used by VoIP SIP: 5060 RTP: 10000–20000 UDP   Configuration 1. Remove SIP Proxy ...

What does DNS Domain and Resolution Failures mean? The Agilico Connect application relies on Domain Name System (DNS) to translate our server addresses into IP addresses. If this test fails, your local device or network is unable to resolve our server names, meaning the application cannot find where to connect. How to fix it: Check Local DNS Setting...

What does SIP ALG Detection mean? SIP Application-Level Gateway (ALG) is a feature on many modern routers designed to help VoIP traffic. Unfortunately, it almost always corrupts the encrypted packets used by modern VoIP systems like Agilico Connect. This results in calls dropping immediately upon answering, phones not ringing, or failure to transfer...

What does this mean? Agilico Connect uses WebSockets (via SignalR) to maintain a persistent, live connection to our servers. This is how the app instantly knows when you are receiving an incoming call or when a colleague's presence status changes to "Busy". If this fails, the app may appear online but won't ring for inbound calls. How to fix it: All...

What does this mean? Agilico Connect requires standard web traffic ports (TCP Port 80 for HTTP and TCP Port 443 for HTTPS) to be open for outbound connections. This is used for downloading configurations, updates, and general API communication. How to fix it: Firewall Rules: Check your local Windows Firewall, antivirus software, or corporate hardwar...

Answering a Call Lift the handset to answer a call, or press the Answer soft key, Speaker, or Headset button.   Placing a Call Lift the handset, press the Speaker key, or press the Headset key.Enter the number, then press Send.   Ending a Call Place the handset on the cradle, press the End Call soft key, or press the Speaker/Headset button.   Mute a...

Overview This guide explains how to configure pfSense firewall for SIP-based VoIP. pfSense is very capable, but defaults like SIP helpers, NAT reflection, and state handling can cause issues if left enabled.   Before you start You’ll need: Web UI access to the firewall Basic understanding of NAT and firewall rules SIP provider info   Ports used by V...

What does STUN Query Blocked mean? Session Traversal Utilities for NAT (STUN) is how the Agilico Connect app discovers your public IP address and the ports opened by your router. This is strictly required for routing live phone calls (audio data) to your headset. How to fix it: Firewall Port Unblocking: Ensure outbound UDP Port 3478 is open on your ...

In this article, we will cover how to generate login details for a user in the Agilico Connect platform and how the user can log in to the Agilico Connect Windows application. The process includes selecting the user within the Connect platform admin console, adding their email address, setting the correct app type, saving the login details and then ...

What does Symmetric NAT mean? For VoIP audio to work correctly, your router must use "Port Preservation" (Consistent NAT). If your router uses "Symmetric NAT" or randomizes the ports it assigns to outbound traffic, Agilico Connect will struggle to establish a stable audio stream, resulting in dead air. How to fix it: Disable Symmetric NAT: Log into ...

What does a NTP Subsystem Timeout mean? Network Time Protocol (NTP) operates on UDP Port 123. It ensures that the clock on your device is perfectly synchronized with global time servers. Secure connections (HTTPS/TLS) will fail if your computer's clock is too far out of sync with our servers servers. How to fix it: Unblock UDP Port 123: Check your f...

What does Double NAT Detected mean? "Double NAT" occurs when you have two routers connected back-to-back (for example, a Wi-Fi mesh node plugged into an ISP router that is not in modem-only mode). This creates two layers of address translation, which heavily disrupts the real-time audio (RTP) required for VoIP calls, leading to one-way audio or drop...

Overview This guide explains how to configure FortiGate firewalls for SIP-based VoIP. FortiGate devices are very capable, but their default SIP handling (session helpers / ALG) often causes issues unless adjusted. Once configured properly, they’re very stable for VoIP.   Before you start You’ll need: Admin access (GUI or CLI) SIP provider details Aw...

This article explains how to reconnect a handset that has lost connection to its base station, or how to pair a new handset to a base station. It applies to Yealink W60B, W70B, W80B and W90B base stations, used with W5x and W7x series handsets. Reconnecting Your Handset (If it stops working) If your phone says “Searching for base” or is not receivin...

Overview This guide covers how to configure TP-Link Omada routers (ER series / Omada SDN) for SIP-based VoIP services. Out of the box, Omada works reasonably well, but there are a couple of settings that can interfere with SIP — mainly SIP ALG and traffic prioritisation. If those aren’t addressed, you may run into issues like one-way audio, unreliab...

Overview This guide covers Fritz!Box routers and SIP-based VoIP. Fritz!Box is designed for consumer VoIP and works well with its own telephony features, but third-party PBX or hosted VoIP can sometimes clash with built-in behaviour.   Before you start Access to Fritz!Box UI SIP provider details   Ports used by VoIP SIP: 5060 RTP: 10000–20000   Confi...

What does this mean? Real-Time Transport Protocol (RTP) carries the actual voice audio during a call. This test simulates a live phone call and measures Packet Loss, Jitter, and Latency. A failure here means your audio will sound robotic, choppy, or delayed. How to fix it: Switch to a Wired Connection: Wi-Fi is highly susceptible to interference and...

Overview This guide covers how to configure DrayTek routers for SIP-based VoIP. DrayTek devices are very commonly used in SMB environments, but they can be a bit sensitive when it comes to SIP — especially around NAT handling and ALG features. A default setup will often “sort of work” but give you intermittent issues like one-way audio or dropped ca...

    Answering a Call Lift the handset Press Speaker for hands‑free Press Headset (if connected)   Placing a Call Dial the desired number Choose how to place the call: Lift handset Press Speaker Press Headset   Ending a Call Hang up the handset Or press the End Call soft key   Muting a Call Press Mute to silence your microphone Press Mute again to un...

To block a phone number, please submit a support ticket, with the number you would like blocked and any relevant details. A member of our support team will then apply the block for you....

Answering a Call Lift the handset Press Speaker for hands‑free Press Headset (if headset mode is enabled)   Placing a Call Using the handset: Lift the handset Dial the number Press Send (if required) Using Speakerphone: Press Speaker Dial the number Press Send Using Headset: Ensure headset mode is active Dial the number Press Send   Ending a Call Ha...

Answering a Call Lift the handset to answer a call, or press the Answer soft key, Speaker, or Headset button.   Placing a Call Lift the handset, press the Speaker key, or press the Headset key.Enter the number, then press Send.   Ending a Call Place the handset on the cradle, press the End Call soft key, or press the Speaker/Headset button.   Mute a...

Answering a Call Lift the handset to answer a call, or press the Answer soft key, Speaker, or Headset button.   Placing a Call Lift the handset, press the Speaker key, or press the Headset key.Enter the number, then press Send.   Ending a Call Hang up the handset, press the End Call soft key, or press Speaker/Headset to disconnect.   Mute a Call Pre...

Answering a Call Lift the handset to answer a call, or press the Answer soft key.   Placing a Call Lift the handset or press the Send/Talk key.Enter the number, then press Send/OK.   Ending a Call Place the handset back in the cradle, press the End Call soft key, or press the On‑hook key.   Mute a Call Press the Mute key to mute the handset micropho...

    Answering a Call Lift the handset for handset operation or press the Speaker for hands‑free operation.You may also press the Headset key if a headset is connected.   Placing a Call Lift the handset, press the Speakerphone button, or press the Headset button.Enter the number, then press Send.   Ending a Call Place the handset on its cradle or pre...

This article provides a step-by-step guide on how to add add a user to a Call Queue or Ring Group. Call Queues and Ring Groups are both tools that help you direct incoming calls to a pre-defined group of people. Ring Group: Everyone in the group is called at the same time. Call Queue: Calls are placed in a line and sent to available team members. Th...

  Answering a Call Lift the handset to answer a call, or press the Answer soft key.   Placing a Call Lift the handset or press the Send/Talk key.Enter the number, then press Send/OK.   Ending a Call Place the handset back in the cradle, press the End Call soft key, or press the On‑hook key.   Mute a Call Press the Mute key to mute the handset microp...

Placing a Call Using the Handset Lift the handset Dial the number Press Send (if required) Using Speakerphone Press Speaker Dial the number Press Send Using a Headset Press Headset Dial the number Press Send   Answering a Call Lift the handset Press Speaker Press Headset Rejecting a call: Press Reject to send it to voicemail.   Ending a Call Hang up...

In this article we will look at how you can create a new pin code for a user of the Agilico Connect Mobile App. Log in to the Agilico Connect Admin Portal ( https://customerportal.hp2k.co.uk/ )   In the left-hand navigation menu, click Users. Locate the required user, then select either the Desktop or Mobile device icon under the Options column   If...