VoIP How To - Page 14

How to dial a number from a web page

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 fo [...]

By |April 6th, 2010|Comments Off on How to dial a number from a web page
The dance of DTMF, SIP & RFC 2833 - An introduction

Edit 16th September 2015: Please note that 3CX Phone System only works with MS Exchange Server 2013 and 2013 SP1 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 [...]

By |March 12th, 2010|Comments Off on The dance of DTMF, SIP & RFC 2833 - An introduction
The main SIP INVITE Header Fields explained

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 [...]

By |March 8th, 2010|Comments Off on The main SIP INVITE Header Fields explained
SIP Messages (Request, Status) Explained

All SIP phone calls consist of messaging for call setup and media for the actual audio. SIP 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 re [...]

By |March 5th, 2010|Comments Off on SIP Messages (Request, Status) Explained
3CX Phone System and ATOM N270 Processor Benchmarking

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 c [...]

By |March 1st, 2010|Comments Off on 3CX Phone System and ATOM N270 Processor Benchmarking
Configuring a CISCO SPA phone to use Active Directory as a phonebook

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 Man [...]

By |February 11th, 2010|Comments Off on Configuring a CISCO SPA phone to use Active Directory as a phonebook
Updating a Beronet BeroFix VoIP Gateway to the latest firmware

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 Fir [...]

By |December 5th, 2009|Comments Off on Updating a Beronet BeroFix VoIP Gateway to the latest firmware
Testing / Configuring a VoIP Provider / SIP Trunk for use with 3CX

Whilst 3CX has a significant number of certified interop partners, in some cases it might be required to work with an untested and/or uncertified VoIP Provider. In these cases you must do a full test run of the VoIP provider to ensure that it will work correctly with the 3CX Phone Sytem. This article explains how. Pre-Requisites You must have at least one (ideally two) external [...]

By |November 5th, 2009|Comments Off on Testing / Configuring a VoIP Provider / SIP Trunk for use with 3CX
Reducing cost with analog phones and an FXS gateway

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 [...]

By |November 4th, 2009|Comments Off on Reducing cost with analog phones and an FXS gateway