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sip

Configuring the Robin Smartview Video Doorphone with 3CX Phone System

February 9, 2012 - 4:47 pm | 5 Comments

If you need a simple SIP-based video door phone, the Robin Smartview should provide a simple solution. This document will explain the steps necessary to set up, configure, and test one of these devices for use with 3CX Phone System.

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Configuring External Extensions with 3CX Phone System

January 15, 2012 - 10:30 am | 10 Comments

Connecting an External Extension to 3CX PhoneSystem is often requested, and can be extremely useful for linking small remote offices and home workers. Here we get a shot at using other phones as External Extensions. Note: If you are looking for instructions to connect a SIP Hardphone using the 3CX Tunnel Protocol via the 3CX SIP Proxy Manager, please view this guide. Using a SIP Hardphone as a Direct External Extension

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VIDEO: General SIP and RTP Overview

May 4, 2011 - 5:31 pm | 0 Comments

Skills Covered by this Module What is SIP What is RTP How calls are setup using SIP Message exchanges

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How to Install and Configure a TFTP Server

March 1, 2011 - 10:12 pm | 2 Comments

TFTP stands for Trivial File Transfer Protocol simply because of its simplicity.  It is mainly used for automated file transfers between machines on UDP Port 69.  In a VoIP environment TFTP is used to upload firmware files to Gateways and Phones and other hardware. Step 1: Download a TFTP Server 1.  You will find pumpkin is an open source TFTP client that is simple to use, follow the link and

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Fax over IP with RelayFax Fax Server and 3CX Phone System

February 17, 2011 - 10:31 pm | 5 Comments

This guide has been tested for RelayFax 7.0. Be aware that previous versions CANNOT work with 3CX PhoneSystem. Note also that more recent versions may have different interface formats and functionality.

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Using a Robin Compact SIP Door Phone with 3CX PhoneSystem

January 27, 2011 - 8:06 pm | 13 Comments

If you need a simple, SIP-based door phone, the Robin Compact SIP C01040 and C01041 provide a simple and inexpensive solution. This document will explain the steps necessary to set up, configure, and test one of these devices. In this example, we will assume that the IP Address of the Robin Compact is 10.0.0.196, and that the IP Address of the 3CX PhoneSystem machine is 10.0.0.11.

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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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VIDEO: Phone Provisioning

April 22, 2010 - 11:06 am | 0 Comments

Skills Covered by this Module Preparing an Extension for Provisioning Advantages of Provisioning through 3CX Management Console The different types of provisioning and what phones they work with Automatic Provisioning using DHCP Option 66 Relevant Course Materials Adjusting The Built-In Phone Provisioning Templates

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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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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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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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Ports used by 3CX Phone System

November 3, 2009 - 7:29 pm | 0 Comments

The following is a complete list of ports that 3CX PhoneSystem uses in a default installation scenario:

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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 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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The Busy Lamp Field (BLF)

September 30, 2009 - 1:44 pm | 0 Comments

Busy Lamp Field (BLF) is a light on an IP phone which tells you whether another extension connected to the same PBX is busy or not. This has to be configured manually from the phone user and it is usually done by making use of the web interface. When configured, the phone subscribes to a resource list available on an IP PBX to be notified with such information about other

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Business Telephone Systems

September 15, 2009 - 3:30 pm | 1 Comments

Five Questions to Help You Decide Which Telephone System Is Best for Your Business There are various options in the market nowadays when it comes to business telephone systems; therefore it can be a challenge to decide which telephone system is best for your organization. In this article we have put together five useful questions that every businessperson should ask him/herself before adopting a new business telephone system. 1. Is

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The STUN Protocol and VoIP – Part 2

September 9, 2009 - 8:15 pm | 1 Comments

As seen in Part 1 of this article, STUN enables a SIP entity running behind a NAT to discover its public IP and what type of NAT is running on the gateway it is connected to. It also enables the SIP entity to discover which port external SIP entities can connect to, to establish a connection. Since SIP uses UDP as transport protocol, and UDP is not a connection oriented

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SNOM 370 Remote Extension. Plug and Play.

April 13, 2009 - 12:51 pm | 3 Comments

I want to start off by giving credit on this nice little piece of engineering to Worksighted systems engineer Matt Scott. I want to start off by defining a remote extension as one that must exit a NAT boundary to traverse the Internet before connecting to the 3CX IP PBX.  This is typically the case when phones are used for remote tele-commuters or in service provider scenarios. Some of you

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Busy Lamp fields? VoIP Article

March 19, 2009 - 8:06 pm | 0 Comments

Busy Lamp fields are the LEDS on phones that indicate the status of other extensions.  BLF fields have been “inherited” from old style PBXs that have this feature. Nowadays, presence information is also displayed via 3CX Assistant, however BLF remains a very popular feature. This article explains more about what BLF is and how it works in SIP context.

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VIDEO: How to setup external extensions using direct SIP

December 18, 2008 - 5:54 pm | 3 Comments

This nugget shows how to setup an external extension using direct SIP. Using the VoIP client as an example it walks you through the process of configuring extensions that reside outside your LAN.

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VIDEO: Firewalls & VoIP

December 4, 2008 - 10:08 pm | 0 Comments

NAT & Firewalls typically present some challenges to VoIP, however with a good understanding of what’s happening it’s possible to make VoIP calls work easily. And being able to have external extensions and bridges with remote offices is one of the cool features of VoIP. This VoIP nugget gives a high level introduction to the problems and what must be done.

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VIDEO: Introduction to SIP & RTP

November 26, 2008 - 9:56 pm | 1 Comments

This video is a high level overview to SIP & RTP – explains the basics of the protocols in a 4 minute video

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