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VoIP How To

Configuring a Draytek 2820 Router for 3CX with QoS configuration

May 24, 2010 - 11:47 am | 9 Comments

In this article we will describe the configuration of a hardware router with a very attractive quality-price ratio. The Draytek 2820 is a ADSL 2/2+ Router/Firewall with advanced features such as the capability to establish 32 VPN connections, a robust object-based firewall, QoS Management, Bandwidth Management, and Dual-WAN cabilities. We will look into NAT configuration for 3CX and QoS configuration to prioritize SIP and RTP traffic. The firmware version tested

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General Configuration Information when using VoIP Providers

May 3, 2010 - 2:20 pm | 3 Comments

This document will provide an overview of the more common concepts which need to be kept in mind when configuring 3CX PhoneSystem inside your network infrastructure, including the following topics: NAT Configuration STUN Resolution DNS Resolution in a VoIP environment Custom Configuration for particular network layouts Different types of VoIP Provider Accounts Custom Source Identification for VoIP Provider SIP message exchange

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Configuring the Portech MV-370 and MV-372

April 26, 2010 - 2:16 pm | 0 Comments

IMPORTANT This device has been tested for Voice using firmware version of “Wed Jun 10 11:53:59 2009”. The guide and template may not work for earlier or later versions of the firmware! Configuration is a 2 step process. First you need to configure 3CX Phone System for use with the Portech. Then, you need to configure the Portech to register and communicate with 3CX PhoneSystem.

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How to dial a number from a web page

April 6, 2010 - 3:45 pm | 0 Comments

Recently Nick published an excellent tip regarding direct dialing from 3CX Assistant on supported phones. I have extended this feature proposing a method to call any number shown on any web page. To make a call from web page via Yealink or Snom phone you need to do following: Use Snom or Yealink phone with latest firmware (important!) You must use Mozilla Firefox (I have not found easy solution for

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Firewall Configuration Overview for 3CX PhoneSystem

March 29, 2010 - 2:57 pm | 0 Comments

In order for 3CX Phone System to communicate successfully with VoIP providers and Remote Extensions, your firewall/router device must be configured for SIP operation. To maximize your chances of success, make sure you choose a device that does not implement a SIP Helper or SIP ALG, or which can be configured to be disabled.

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BeroNet bero*fix Box Configuration

March 18, 2010 - 7:13 pm | 0 Comments

This guide will show you how to configure a Beronet bero*fix box. The main target of this guide is to connect the bero*fix card with the LAN and WAN. . . Setting up the BeroNet bero*fix 400/1600/6400 Connect the box to the network and plug in power supply The default IP address of the bero*fix  is 10.0.0.2/16.  If your LAN operates in the same network make sure that the IP

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Configure BeroNet bero*fix CARD via Network bridge

March 18, 2010 - 7:12 pm | 0 Comments

This guide will show you how to configure a Beronet bero*fix card. The main scope of this guide is to connect the bero*fix card via a network brigde with the LAN and WAN. Setting up the BeroNet bero*fix 400/1600/6400 Insert the Beronet Berofix into the PC After the system has booted you will have a new network device available Windows usually detects the card as a “Realtek RTL8139” If you

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Introduction to DTMF & SIP & RFC 2833

March 12, 2010 - 2:45 pm | 0 Comments

Introduction DTMF (dual tone multi frequency) is the signal to the phone company that you generate when you press an ordinary telephone’s touch keys. In the United States, it’s known as a “Touchtone” phone. DTMF has generally replaced loop disconnect (“pulse”) dialling. With DTMF, each key you press on your phone generates two tones of specific frequencies. So that a voice can’t imitate the tones, one tone is generated from

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ISDN Outbound caller ID and how to reformat it on a BeroFix

March 8, 2010 - 8:10 pm | 0 Comments

This guide describes the different caller ID formats and how to alter /rewrite the caller ID before it gets sent to the Telecoms provider when using beroFix VoIP Gateway cards. In this document we will use a German number (National code +49) from Dusseldorf (area code 0211) as an example how to format the caller ID: A telephone line-number (TelN) 12345600 (where ’00′ are the optional DIDs) A DID to

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The main SIP INVITE Header Fields explained

March 8, 2010 - 4:15 pm | 0 Comments

This article explains the main fields included in a SIP INVITE, which is sent to set-up a VoIP call. A SIP INVITE message contains typically between 4 and 6 header entries with contact information inside them. Different devices or providers use these headers in different ways and therefore, an understanding of the “normal” scope of these headers, together with a close analysis of log messages, will help to understand the

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SIP messages explained

March 5, 2010 - 4:17 pm | 1 Comments

All SIP Phone calls consist of 2 channels: Messaging – SIP Media – RTP Messaging is carried using UDP or TCP (normally UDP) SIP packets – It is used for device registration, call setup/teardown, on-hold notification, call transfers, DTMF signals. Typically the PBX Server is expecting to receive SIP messages on port 5060, so the PBX Server must be able to receive these packets, requiring accurate firewall configuration to allow

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How to configure Secure SiP – TLS

March 4, 2010 - 2:53 pm | 3 Comments

This is a complete guide that will provide detailed information on how to: 1. Setup 3CX Phone System with Secure SiP (TLS) (In this way the SiP messaging will be encrypted and therefore more secure) 2. Create certificates with Simple CA. 3. Use Microsoft’s inbuilt importer for trusted certificates and 4. Configure IPPhones to communicate SiP securely. Phones mentioned in this article will be 3CXPhone, Eyebeam and Snom

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3CX Phone System and ATOM N270 Processor Benchmarking

March 1, 2010 - 4:22 pm | 8 Comments

Many businesses who are adopting VoIP Software PBX Systems are always looking for a way to cut down costs. That’s where the Intel Atom N270 comes in the picture, which is  designed to deliver performance with minimal power usage. In addition many of our partners have requested the ability to put together a low cost dedicated server based on an Intel Atom.  The Intel Atom N270 can be found in

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Configuring a CISCO SPA phone to use Active Directory as a phonebook

February 11, 2010 - 4:49 pm | 0 Comments

Cisco SPA Phones have the ability to look up for phone numbers and contacts in an Active Directory/LDAP. For this a template needs to be created and  the Cisco Phones need to be re-provisioned with this template to enable them  to access Active Directory and take AD user information from there. Step 1: Locating the Cisco template in the 3CX Management Console Access the 3CX Management Console and click on

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Microsoft IIS or Ultidev Cassini – Which do you use?

January 22, 2010 - 1:39 pm | 5 Comments

Poll on which webserver is used most with 3CX

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3rd Party Supported VoIP Providers

January 1, 2010 - 9:00 am | 0 Comments

The VOIP Providers listed below are not directly available in the ‘Add VOIP’ section of the 3CX Phone System Web Interface, but are known to work with 3CX Phone System and tested by 3rd parties.

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Updated IP phone configuration guides

December 21, 2009 - 4:20 pm | 0 Comments

Updated IP phone configuration guides for the most popular phones supported by 3CX

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Updating a Beronet BeroFix VoIP Gateway to the latest firmware

December 5, 2009 - 8:11 pm | 0 Comments

If you are not running the latest version of the firmware, please update your Beronet device firmware to the recommended version as explained below. Update Procedure To update go to http://www.beronet.com/downloads/berofix/ and download the appropriate update. Log on to the beroFix Webinterface, click on Preferences / Update Tool or Management / Update Tool depending on the Firmware Reboot the device after updating

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Configuring and testing a VoIP Provider for use with 3CX PhoneSystem

November 5, 2009 - 7:04 pm | 0 Comments

Note 1: The results generated from these tests should remain valid while the VoIP Provider keeps your account settings static. If your VoIP Provider account is reconfigured, it may be necessary to redo these tests. Note 2: The supported configuration for interoperability with VoIP Providers is for the PBX to be behind a WAN-to-LAN device, with Port Forwarding implemented correctly.

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Reducing cost with analog phones and an FXS gateway

November 4, 2009 - 5:54 pm | 3 Comments

When deploying a new VoIP phone system, especially in larger installs, the cost of new IP phones can be substantial. And for many work stations, the current analog phone might do ‘just fine’. And with 3CX Assistant, a desktop call control utility for Windows, extension users can enjoy all the benefits of an IP phone including presence and call control, whilst using their analog phone. Additionally, in environments such as

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Direct SIP Explained

November 3, 2009 - 12:53 pm | 0 Comments

How does direct SIP work? The way direct SIP calls are initiated is very similar to the way sending an email works. A caller can call a callee from either a stand alone SIP phone which is not registered to any SIP server,  to a VoIP provider, or from a SIP phone registered to an IP PBX, which supports direct SIP. If a user calls “robert@voipproducts.org” from a stand alone

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The top 10 advantages of a Windows based PBX

November 2, 2009 - 3:45 pm | 0 Comments

Why your next Phone System should be software based This whitepaper outlines the top 10 advantages of choosing a Windows based PBX as your next phone system. Why it makes sense to move away from proprietary solutions such as Nortel, Mitel, Avaya, Alcatel and Siemens, and go straight to a REAL software based solution. And why it should be running on Microsoft Windows! Click here to read the whitepaper or

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Voip Security

October 24, 2009 - 6:34 pm | 0 Comments

Ten tips for a more secure IP PBX System If you are in charge of a VoIP system or planning to deploy one, then VoIP security is probably one of the considerations. In this article we present ten simple tips that could help make an IP PBX System more resilient to network attacks. Protect the administrative interface by using a strong password – Most modern IP PBX servers can be

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The Session Description Protocol (SDP)-VoIP Part 2

October 11, 2009 - 7:49 pm | 1 Comments

When a SIP based VoIP call is established, the audio or video sent between two SIP entities or more is streamed. Since many different codecs are supported by different devices or software, and each individual SIP entity taking part in the call does not know the IP address of the other SIP entity or to which port the stream should be sent to, SDP is used to advertise such details

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The Session Description Protocol (SDP)-VoIP Part 1

October 11, 2009 - 7:09 pm | 0 Comments

When media is streamed in a SIP based voice over IP call, being audio, or video or both, one of the requirements is for the participants to know the media details, i.e. transport address, transport protocol, codec, ports and other session description metadata. SDP, also known as Session Description Protocol is the protocol used with SIP (session initiation protocol) to advertise such information. SDP does not stream or provide the

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