Yealink SIP-T4X, SIP-T42G, SIP-T41P Administration Manual

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T4X IP Phone

Administrator Guide

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THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS GUIDE ARE

SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND

RECOMMENDATIONS IN THIS GUIDE ARE BELIEVED TO BE ACCURATE AND PRESENTED

WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL

RESPONSIBILITY FOR THEIR APPLICATION OF PRODUCTS.

YEALINK NETWORK TECHNOLOGY CO., LTD. MAKES NO WARRANTY OF ANY KIND WITH

REGARD TO THIS GUIDE, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF

MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. Yealink Network Technology

CO., LTD. shall not be liable for errors contained herein nor for incidental or consequential

damages in connection with the furnishing, performance, or use of this guide.

Hereby, Yealink Network Technology CO., LTD. declares that this phone is in conformity

with the essential requirements and other relevant provisions of the CE, FCC.

This device is marked with the CE mark in compliance with EC Directives 2006/95/EC and 2004/108/EC.

This device is compliant with Part 15 of the FCC Rules. Operation is subject to the following two conditions:

1. This device may not cause harmful interference, and

2. This device must accept any interference received, including interference that may cause undesired

operation.

Note: This device is tested and complies with the limits for a Class B digital device, pursuant to Part 15 of the

FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a

residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not

installed and used in accordance with the instructions, may cause harmful interference to radio

communications. However, there is no guarantee that interference will not occur in a particular installation. If

this equipment does cause harmful interference to radio or television reception, which can be determined

by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more

of the following measures:

1. Reorient or relocate the receiving antenna.

2. Increase the separation between the equipment and receiver.

3. Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.

4. Consult the dealer or an experience radio/TV technician for help.

To avoid the potential effects on the environment and human health as a result of the

presence of hazardous substances in electrical and electronic equipment, end users of

electrical and electronic equipment should understand the meaning of the crossed-out

wheeled bin symbol. Do not dispose of WEEE as unsorted municipal waste and have to

collect such WEEE separately.

We are striving to improve our documentation quality and we appreciate your feedback. Email

your opinions and comments to DocsFeedback@yealink.com.

Yealink SIP-T4X IP phone firmware contains third-party software under the GNU General Public License

(GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms

and conditions of the license.

The original GPL license, source code of components licensed under GPL and used in Yealink products can

be downloaded online:

http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293.

About This Guide

The guide is intended for administrators who need to properly configure, customize,

manage, and troubleshoot the IP phone system rather than the end-users. It provides

details on the functionality and configuration of SIP-T4X IP phones.

Many of the features described in this guide involve network settings, which could affect

the IP phone’s performance in the network. So an understanding of the IP networking

and prior knowledge of IP telephony concepts are necessary.

This guide covers SIP-T46G, T42G and T41P IP phones. The following related documents

for SIP-T4X IP phones are available:

 Quick Installation Guides, which describe how to assemble IP phones.

 Quick Reference Guides, which describe the most basic features available on IP

phones.

 User Guides, which describe basic and advanced features available on IP phones.

 Auto Provisioning Deployment Guide, which describes how to provision IP phones

using the configuration files.

 Configuration Conversion Tool User Guide, which describes how to convert and

encrypt the configuration files using the Configuration Conversion Tool.

 <y0000000000xx>.cfg and <MAC>.cfg template configuration files.

 IP Phones Deployment Guide for BroadSoft UC-One Environments, which describes

how to configure the BroadSoft features on the BroadWorks web portal and IP

phones.

For support or service, please contact your Yealink reseller or go to Yealink Technical

Support online: http://www.yealink.com/Support.aspx.

The information detailed in this guide is applicable to the firmware version 72 or higher.

The firmware format is like x.x.x.x.rom. The second x from left should be greater than or

equal to 72 (e.g., the firmware version of SIP-T46G IP phone: 28.72.0.10.rom). This

administrator guide includes the following chapters:

 Chapter 1, “Product Overview” describes SIP components and SIP IP phones.

 Chapter 2, “Getting Started” describes how to install and connect IP phones and

configuration methods.

Administrator’s Guide for SIP-T4X IP Phones

 Chapter 3, “Configuring Basic Features” describes how to configure basic features

on IP phones.

 Chapter 4, “Configuring Advanced Features” describes how to configure

advanced features on IP phones.

 Chapter 5, “Configuring Audio Features” describes how to configure audio features

on IP phones.

 Chapter 6, “Configuring Security Features” describes how to configure security

features on IP phones.

 Chapter 7, “Upgrading Firmware” describes how to upgrade the firmware of IP

phones.

 Chapter 8, “Resource Files” describes the resource files that can be downloaded

by IP phones.

 Chapter 9, “Troubleshooting” describes how to troubleshoot IP phones and

provides some common troubleshooting solutions.

 Chapter 10, “Appendix” provides the glossary, reference information about IP

phones compliant with RFC 3261, SIP call flows and sample configuration files.

This section describes the changes to this guide for each release and guide version.

The following sections are new for this version:

 Power Indicator LED on page 36

 Contrast on page 38

Major updates have occurred to the following sections:

 DHCP on page 18

 Backlight on page 41

 Time and Date on page 47

 Key as Send on page 59

 Anonymous Call on page 75

 Busy Lamp Field on page 136

 Action URL on page 156

 IPv6 Support on page 187

 Transport Layer Security on page 203

About This Guide

vii

Major updates have occurred to the following section:

 Language on page 52

Major updates have occurred to the following sections:

 Language on page 52

 Anonymous Call on page 75

Major updates have occurred to the following sections:

 Backlight on page 41

 Language on page 52

 Logo Customization on page 55

 Anonymous Call on page 75

 Action URL on page 156

 Action URI on page 159

 Audio Codecs on page 193

Major updates have occurred to the following sections:

 Language on page 52

 Auto Answer on page 72

 Audio Codecs on page 193

 Encrypting Configuration Files on page 211

This version is updated to incorporate T41P as one of the T4X device models. The

following section is new for this version:

 Logo Customization on page 55

Administrator’s Guide for SIP-T4X IP Phones

viii

Major updates have occurred to the following sections:

 SIP IP Phone Models on page 3

 Configuring Transmission Methods of the Internet Port and PC Port on page 25

 Language on page 52

 Remote Phone Book on page 131

 Server Redundancy on page 162

 Audio Codecs on page 193

 Transport Layer Security on page 203

 Secure Real-Time Transport Protocol on page 209

This version is updated to incorporate T42G as one of the T4X device models. The

following section is new for this version:

 SIP IP Phone Models on page 3

Major updates have occurred to the following sections:

 Reading Icons on page 16

 PPPoE on page 24

 Backlight on page 41

 Language on page 52

 Call Completion on page 74

 TR-069 Device Management on page 185

 IPv6 Support on page 187

 Audio Codecs on page 193

 Upgrading Firmware on page 215

 Configuring DSS Key on page 381

Table of Contents

About This Guide ...................................................................... v

Documentations ............................................................................................................................... v

In This Guide .................................................................................................................................... v

Summary of Changes .................................................................................................................... vi

Changes for Release 72, Guide Version 72.1 ........................................................................ vi

Changes for Release 71.0, Guide Version 71.181 ................................................................ vii

Changes for Release 71.0, Guide Version 71.180 ................................................................ vii

Changes for Release 71.0, Guide Version 71.171 ................................................................ vii

Changes for Release 71.0, Guide Version 71.170 ................................................................ vii

Changes for Release 71.0, Guide Version 71.150 ................................................................ vii

Changes for Release 71.0, Guide Version 71.80 ................................................................. viii

Table of Contents .................................................................... ix

Product Overview ..................................................................... 1

VoIP Principle .................................................................................................................................... 1

SIP Components............................................................................................................................... 2

SIP IP Phone Models ........................................................................................................................ 3

Physical Features of SIP-T4X IP Phones ................................................................................... 4

Key Features of SIP-T4X IP Phones ........................................................................................... 6

Getting Started ......................................................................... 9

Connecting the IP Phone ................................................................................................................. 9

Initialization Process Overview .................................................................................................... 12

Verifying Startup ............................................................................................................................ 14

Configuration Methods ................................................................................................................. 14

Phone User Interface.............................................................................................................. 14

Web User Interface ................................................................................................................ 14

Configuration Files.................................................................................................................. 15

Reading Icons ................................................................................................................................ 16

Configuring Basic Network Parameters ...................................................................................... 18

DHCP ....................................................................................................................................... 18

Configuring Network Parameters Manually ........................................................................ 21

PPPoE ....................................................................................................................................... 24

Configuring Transmission Methods of the Internet Port and PC Port ................................. 25

Creating Dial Plan ......................................................................................................................... 28

Administrator’s Guide for SIP-T4X IP Phones

Replace Rule ........................................................................................................................... 29

Dial-now .................................................................................................................................. 30

Area Code............................................................................................................................... 32

Block Out ................................................................................................................................. 33

Configuring Basic Features .................................................... 35

Power Indicator LED ...................................................................................................................... 36

Contrast .......................................................................................................................................... 38

Wallpaper ....................................................................................................................................... 39

Backlight ......................................................................................................................................... 41

User Password ............................................................................................................................... 42

Administrator Password ................................................................................................................ 43

Phone Lock ..................................................................................................................................... 44

Time and Date ............................................................................................................................... 47

Language ....................................................................................................................................... 52

Loading Language Packs ...................................................................................................... 53

Specifying the Language to Use........................................................................................... 54

Logo Customization ....................................................................................................................... 55

Softkey Layout................................................................................................................................ 56

Key as Send ................................................................................................................................... 59

Hotline ............................................................................................................................................ 61

Call Log ........................................................................................................................................... 63

Missed Call Log ............................................................................................................................. 64

Local Directory ............................................................................................................................... 65

Live Dialpad ................................................................................................................................... 68

Call Waiting .................................................................................................................................... 68

Auto Redial ..................................................................................................................................... 70

Auto Answer ................................................................................................................................... 72

Call Completion ............................................................................................................................. 74

Anonymous Call ............................................................................................................................. 75

Anonymous Call Rejection ............................................................................................................ 77

Do Not Disturb ................................................................................................................................ 79

Busy Tone Delay ............................................................................................................................. 83

Return Code When Refuse ............................................................................................................ 84

Early Media .................................................................................................................................... 85

180 Ring Workaround .................................................................................................................... 86

Use Outbound Proxy in Dialog ..................................................................................................... 87

SIP Session Timer ........................................................................................................................... 88

Session Timer ................................................................................................................................. 89

Call Hold ......................................................................................................................................... 90

Call Forward .................................................................................................................................. 92

Call Transfer ................................................................................................................................... 98

Network Conference ................................................................................................................... 100

Transfer on Conference Hang Up .............................................................................................. 101

Table of Contents

Directed Call Pickup .................................................................................................................... 102

Group Call Pickup ........................................................................................................................ 105

Dialog-Info Call Pickup ................................................................................................................ 108

Call Return .................................................................................................................................... 110

Call Park ....................................................................................................................................... 111

Web Server Type.......................................................................................................................... 112

Calling Line Identification Presentation ..................................................................................... 114

Connected Line Identification Presentation .............................................................................. 115

DTMF ............................................................................................................................................. 116

Suppress DTMF Display .............................................................................................................. 119

Transfer via DTMF ........................................................................................................................ 120

Intercom ........................................................................................................................................ 121

Outgoing Intercom Calls ...................................................................................................... 121

Incoming Intercom Calls ...................................................................................................... 122

Configuring Advanced Features...........................................125

Distinctive Ring Tones .................................................................................................................. 125

Tones ............................................................................................................................................. 129

Remote Phone Book .................................................................................................................... 131

LDAP .............................................................................................................................................. 133

Busy Lamp Field ........................................................................................................................... 136

Music on Hold .............................................................................................................................. 141

Automatic Call Distribution ......................................................................................................... 142

Message Waiting Indicator ........................................................................................................ 144

Multicast Paging .......................................................................................................................... 147

Sending RTP Stream ............................................................................................................. 147

Receiving RTP Stream .......................................................................................................... 149

Call Recording ............................................................................................................................. 151

Hot Desking .................................................................................................................................. 155

Action URL .................................................................................................................................... 156

Action URI ..................................................................................................................................... 159

Server Redundancy ..................................................................................................................... 162

SIP Server Domain Name Resolution .................................................................................. 164

LLDP ............................................................................................................................................... 167

VLAN ............................................................................................................................................. 170

VPN ................................................................................................................................................ 174

Quality of Service ........................................................................................................................ 176

Network Address Translation ..................................................................................................... 178

802.1X Authentication ................................................................................................................. 180

TR-069 Device Management ...................................................................................................... 185

IPv6 Support ................................................................................................................................. 187

Configuring Audio Features ..................................................191

Administrator’s Guide for SIP-T4X IP Phones

xii

Headset Prior ............................................................................................................................... 191

Dual Headset ............................................................................................................................... 192

Audio Codecs .............................................................................................................................. 193

Acoustic Clarity Technology ........................................................................................................ 197

Acoustic Echo Cancellation ................................................................................................. 197

Voice Activity Detection ....................................................................................................... 198

Comfort Noise Generation .................................................................................................. 199

Jitter Buffer ............................................................................................................................ 200

Configuring Security Features ...............................................203

Transport Layer Security .............................................................................................................. 203

Secure Real-Time Transport Protocol .......................................................................................... 209

Encrypting Configuration Files ................................................................................................... 211

Upgrading Firmware .............................................................215

Resource Files ........................................................................219

Replace Rule Template ............................................................................................................... 219

Dial-now Template ....................................................................................................................... 220

Softkey Layout Template ............................................................................................................. 221

Local Contact File ........................................................................................................................ 223

Remote XML Phone Book ............................................................................................................ 224

Directory Template ...................................................................................................................... 226

Super Search Template ............................................................................................................... 227

Specifying the Access URL of Resource Files ............................................................................ 228

Troubleshooting .....................................................................231

Troubleshooting Methods ........................................................................................................... 231

Viewing Log Files .................................................................................................................. 231

Capturing Packets ................................................................................................................ 234

Enabling the Watch Dog Feature ........................................................................................ 234

Getting Information from Status Indicators ........................................................................ 235

Analyzing Configuration Files ............................................................................................. 235

Troubleshooting Solutions ........................................................................................................... 236

Why is the LCD screen blank? ............................................................................................. 236

Why doesn’t the IP phone get an IP address? ................................................................... 236

How do I find the basic information of the IP phone? ....................................................... 237

Why doesn’t the IP phone upgrade firmware successfully? ............................................. 237

Why doesn’t the IP phone display time and date correctly? ........................................... 237

Why do I get poor sound quality during a call? ................................................................ 237

What is the difference between a remote phone book and a local phonebook? ........ 238

What is the difference between user name, register name and display name? .......... 238

Table of Contents

xiii

How to reboot IP phone remotely? ..................................................................................... 238

How to increase or decrease the volume? ........................................................................ 238

What will happen if I connect both PoE cable and power adapter? Which has the higher

priority? .................................................................................................................................. 239

What is auto provisioning? .................................................................................................. 239

What is PnP? .......................................................................................................................... 239

Why doesn’t the IP phone update the configuration? ...................................................... 239

What do “on code” and “off code” mean? ....................................................................... 239

How to solve the IP conflict problem? ................................................................................ 240

How to reset your phone to factory configurations? ......................................................... 240

How to restore the administrator password? .................................................................... 241

Appendix ...............................................................................243

Appendix A: Glossary ................................................................................................................. 243

Appendix B: Time Zones ............................................................................................................. 245

Appendix C: Configuration Parameters .................................................................................... 248

Setting Parameters in Configuration Files .......................................................................... 248

Basic and Advanced Feature Parameters ......................................................................... 248

Audio Feature Parameters ................................................................................................... 359

Upgrading Firmware ........................................................................................................... 371

Resource Files ....................................................................................................................... 374

Troubleshooting .................................................................................................................... 379

Configuring DSS Key ............................................................................................................ 381

Appendix D: SIP (Session Initiation Protocol) ............................................................................ 396

RFC and Internet Draft Support .......................................................................................... 396

SIP Request ............................................................................................................................ 399

SIP Header ............................................................................................................................ 400

SIP Responses ....................................................................................................................... 401

SIP Session Description Protocol (SDP) Usage .................................................................. 403

Appendix E: SIP Call Flows ......................................................................................................... 404

Successful Call Setup and Disconnect ............................................................................... 405

Unsuccessful Call Setup—Called User is Busy .................................................................. 407

Unsuccessful Call Setup—Called User Does Not Answer ................................................ 410

Successful Call Setup and Call Hold .................................................................................. 412

Successful Call Setup and Call Waiting ............................................................................. 414

Call Transfer without Consultation ...................................................................................... 419

Call Transfer with Consultation ............................................................................................ 423

Always Call Forward ............................................................................................................ 428

Busy Call Forward ................................................................................................................ 432

No Answer Call Forward ..................................................................................................... 435

Call Conference .................................................................................................................... 438

Appendix F: Sample Configuration File .................................................................................... 443

Index ......................................................................................449

Administrator’s Guide for SIP-T4X IP Phones

xiv

Product Overview

This chapter contains the following information about SIP-T4X IP phones:

 VoIP Principle

 SIP Components

 SIP IP Phone Models

VoIP

VoIP (Voice over Internet Protocol) is a technology using the Internet Protocol instead of

traditional Public Switch Telephone Network (PSTN) technology for voice

communications.

It is a family of technologies, methodologies, communication protocols, and

transmission techniques for the delivery of voice communications and multimedia

sessions over IP networks. The H.323 and Session Initiation Protocol (SIP) are two

popular VoIP protocols that are found in widespread implementation.

H.323

H.323 is a recommendation from the ITU Telecommunication Standardization Sector

(ITU-T) that defines the protocols to provide audio-visual communication sessions on

any packet network. The H.323 standard addresses call signaling and control,

multimedia transport and control, and bandwidth control for point-to-point and

multi-point conferences.

It is widely implemented by voice and video conference equipment manufacturers, is

used within various Internet real-time applications such as GnuGK and NetMeeting and

is widely deployed by service providers and enterprises for both voice and video

services over IP networks.

SIP

SIP (Session Initiation Protocol) is the Internet Engineering Task Force’s (IETF’s) standard

for multimedia conferencing over IP. It is an ASCII-based, application-layer control

protocol (defined in RFC 3261) that can be used to establish, maintain, and terminate

calls between two or more endpoints. Like other VoIP protocols, SIP is designed to

address the functions of signaling and session management within a packet telephony

network. Signaling allows call information to be carried across network boundaries.

Session management provides the ability to control the attributes of an end-to-end call.

Administrator’s Guide for SIP-T4X IP Phones

SIP provides capabilities to:

 Determine the location of the target endpoint -- SIP supports address resolution,

name mapping, and call redirection.

 Determine the media capabilities of the target endpoint -- Via Session Description

Protocol (SDP), SIP determines the “lowest level” of common services between

endpoints. Conferences are established using only the media capabilities that can

be supported by all endpoints.

 Determine the availability of the target endpoint -- A call cannot be completed

because the target endpoint is unavailable. SIP determines whether the called

party is already on the IP phone or does not answer in the allotted number of rings.

It then returns a message indicating why the target endpoint is unavailable.

 Establish a session between the origin and target endpoint -- The call can be

completed, SIP establishes a session between endpoints. SIP also supports mid-call

changes, such as the addition of another endpoint to the conference or the change

of a media characteristic or codec.

 Handle the transfer and termination of calls -- SIP supports the transfer of calls from

one endpoint to another. During a call transfer, SIP simply establishes a session

between the transferee and a new endpoint (specified by the transferring party)

and terminates the session between the transferee and the transferring party. At

the end of a call, SIP terminates the sessions between all parties.

SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A

user agent can function as one of the following roles:

 User Agent Client (UAC) -- A client application that initiates the SIP request.

 User Agent Server (UAS) -- A server application that contacts the user when a SIP

request is received and that returns a response on behalf of the user.

User Agent Client (UAC)

The UAC is an application that initiates up to six feasible SIP requests to the UAS. The six

requests issued by the UAC are: INVITE, ACK, OPTIONS, BYE, CANCEL and REGISTER.

When the SIP session is being initiated by the UAC SIP component, the UAC determines

the information essential for the request, which is the protocol, the port and the IP

address of the UAS to which the request is being sent. This information can be dynamic

and will make it challenging to put through a firewall. For this reason, it may be

recommended to open the specific application type on the firewall. The UAC is also

capable of using the information in the request URI to establish the course of the SIP

request to its destination, as the request URI always specifies the host which is essential.

The port and protocol are not always specified by the request URI. Thus if the request

does not specify a port or protocol, a default port or protocol is contacted. It may be

Product Overview

preferential to use this method when not using an application layer firewall. Application

layer firewalls like to know what applications are flowing though which ports and it is

possible to use content types of other applications other than the one you are trying to

let through what has been denied.

User agent server (UAS)

UAS is a server that hosts the application responsible for receiving the SIP requests from

a UAC, and on reception it returns a response to the request back to the UAC. The UAS

may issue multiple responses to the UAC, not necessarily a single response.

Communication between UAC and UAS is client/server and peer-to–peer.

Typically, a SIP endpoint is capable of functioning as both a UAC and a UAS, but it

functions only as one or the other per transaction. Whether the endpoint functions as a

UAC or a UAS depends on the UA that initiates the request.

This section introduces the SIP-T4X IP phone family. SIP-T4X IP phones are endpoints in

the overall network topology, which are designed to interoperate with other compatible

equipments including application servers, media servers, internet-working gateways,

voice bridges, and other endpoints. SIP-T4X IP phones are characterized by a large

number of functions, which simplify business communication with a high standard of

security and can work seamlessly with a large number of SIP PBXs.

SIP-T4X IP phones provide a powerful and flexible IP communication solution for Ethernet

TCP/IP networks, delivering excellent voice quality. The high-resolution graphic display

supplies content in multiple languages for system status, call log and directory access.

SIP-T4X IP phones also support advanced functionalities, including LDAP, Busy Lamp

Field, Sever Redundancy and Network Conference.

The following IP phone models are described:

 SIP-T46G

 SIP-T42G

 SIP-T41P

SIP-T4X IP phones comply with the SIP standard (RFC 3261), and they can only be used

within a network that supports this type of phone.

In order to operate as SIP endpoints in your network successfully, SIP-T4X IP phones must

meet the following requirements:

 A working IP network is established.

 Routers are configured for VoIP.

 VoIP gateways are configured for SIP.

 The latest (or compatible) firmware of SIP-T4X IP phones is available.

Administrator’s Guide for SIP-T4X IP Phones

 A call server is active and configured to receive and send SIP messages.

This section lists the available physical features of SIP-T4X IP phones.

SIP-T46G

Physical Features:

- 4.3” TFT-LCD, 480 x 272 pixel, 16.7M colors

- 6 VoIP accounts, BroadSoft/Avaya/Asterisk validated

- HD Voice: HD Codec, HD Handset, HD Speaker

- 41 keys including 10 line keys

- 1xRJ9 (4P4C) handset port

- 1xRJ9 (4P4C) headset port

- 2xRJ45 10/100/1000Mbps Ethernet ports

- 1XRJ12 (6P6C) expansion module port

- 14 LEDs: 1xpower, 10xline, 1xmute, 1xheadset, 1xspeakerphone

- Power adapter: AC 100~240V input and DC 5V/2A output

- Power over Ethernet (IEEE 802.3af)

- Built-in USB port, support Bluetooth headset

Product Overview

SIP-T42G

Physical Features:

- 192 x 64 graphic LCD

- 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated

- HD Voice: HD Codec, HD Handset, HD Speaker

- 35 keys including 6 line keys

- 1xRJ9 (4P4C) handset port

- 1xRJ9 (4P4C) headset port

- 2xRJ45 10/100/1000Mbps Ethernet ports

- 1XRJ12 (6P6C) EHS36 headset adapter port

- 10 LEDs: 1xpower, 6xline, 1xmute, 1xheadset, 1xspeakerphone

- Power adapter: AC 100~240V input and DC 5V/1.2A output

- Power over Ethernet (IEEE 802.3af)

Administrator’s Guide for SIP-T4X IP Phones

SIP-T41P

Physical Features:

- 192 x 64 graphic LCD

- 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated

- HD Voice: HD Codec, HD Handset, HD Speaker

- 35 keys including 6 line keys

- 1xRJ9 (4P4C) handset port

- 1xRJ9 (4P4C) headset port

- 2xRJ45 10/100Mbps Ethernet ports

- 1XRJ12 (6P6C) EHS36 headset adapter port

- 10 LEDs: 1xpower, 6xline, 1xmute, 1xheadset, 1xspeakerphone

- Power adapter: AC 100~240V input and DC 5V/1.2A output

- Power over Ethernet (IEEE 802.3af)

In addition to physical features introduced above, SIP-T4X IP phones also support the

following key features when running the latest firmware:

 Phone Features

- Call Options: emergency call, call waiting, call hold, call mute, call forward,

call transfer, call pickup, conference.

- Basic Features: DND, phone lock, auto redial, live dialpad, dial plan, hotline,

Product Overview

caller identity, auto answer.

- Advanced Features: BLF, server redundancy, distinctive ring tones, remote

phone book, LDAP, 802.1x authentication.

 Codecs and Voice Features

- Wideband codec: G.722

- Narrowband codec: G.711, G.723.1, G.726, G.729AB, GSM, iLBC .

- VAD, CNG, AEC, PLC, AJB, AGC

- Full-duplex speakerphone with AEC

 Network Features

- SIP v1 (RFC2543), v2 (RFC3261)

- IPv4/IPv6 support

- NAT Traversal: STUN mode

- DTMF: INBAND, RFC2833, SIP INFO

- Proxy mode and peer-to-peer SIP link mode

- IP assignment: Static/DHCP/PPPoE (for SIP-T46G only)

- TFTP/DHCP client

- HTTP/HTTPS server

- DNS client

- NAT/DHCP server

 Management

- FTP/TFTP/HTTP/PnP auto-provision

- Configuration: browser/phone/auto-provision

- Direct IP call without SIP proxy

- Dial number via SIP server

- Dial URL via SIP server

 Security

- HTTPS (server/client)

- SRTP (RFC3711)

- Transport Layer Security (TLS)

- VLAN (802.1q), QoS

- Digest authentication using MD5/MD5-sess

- Secure configuration file via AES encryption

- Phone lock for personal privacy protection

- Admin/User configuration mode

Administrator’s Guide for SIP-T4X IP Phones

Getting Started

This chapter provides basic information and installation instructions of SIP-T4X IP phones.

This chapter provides the following sections:

 Connecting the IP Phone

 Initialization Process Overview

 Verifying Startup

 Configuration Methods

 Reading Icons

 Configuring Basic Network Parameters

 Creating Dial Plan

This section introduces how to install SIP-T4X IP phones with the components in

packaging contents.

1. Attach the stand

2. Connect the handset and optional headset

3. Connect the network and power

Note

A headset, wall mount bracket and power adapter are not included in packaging contents.

Administrator’s Guide for SIP-T4X IP Phones

1) Attach the stand:

Desk Mount Method

Wall Mount Method (Optional)

Note

For more information on how to mount the phone to a wall, refer to

Yealink Wall Mount

Quick Installation Guide for SIP-T4X IP Phones

Getting Started

2) Connect the handset, optional headset and Bluetooth headset:

Note

3) Connect the network and power:

 AC power

 Power over Ethernet (PoE)

AC Power

To connect the AC power and network:

1. Connect the DC plug of the power adapter to the DC5V port on IP phones and

connect the other end of the power adapter into an electrical power outlet.

2. Connect the included or a standard Ethernet cable between the Internet port on IP

phones and the one on the wall or switch/hub device port.

Wireless headset adapter EHS36 and Bluetooth USB dongle should be purchased separately.

For more information on how to use the EHS36 on the IP phone, refer to

Yealink EHS36

User Guide.

Bluetooth can only be used on SIP-T46G IP phones. For more information on how to use the Bluetooth on SIP-T46G IP phones, refer to

Yealink Bluetooth USB Dongle BT40 User

Guide

EXT port can also be used to connect the expansion module EXP40. For more information on how to connect EXP40, refer to

Yealink EXP40 User Guide

Administrator’s Guide for SIP-T4X IP Phones

Power over Ethernet

With the included or a regular Ethernet cable, IP phones can be powered from a

PoE-compliant switch or hub.

To connect the PoE:

1. Connect the Ethernet cable between the Internet port on IP phones and an

available port on the in-line power switch/hub.

Note

The initialization process of IP phones is responsible for network connectivity and

operation of IP phones in your local network.

Once you connect your IP phone to the network and to an electrical supply, the IP phone

begins its initialization process.

During the initialization process, the following events proceed:

Loading the ROM file

The ROM file resides in the flash memory of IP phones. IP phones come from the factory

with a ROM file preloaded. During initialization, IP phones run a bootstrap loader that

loads and executes the ROM file.

Configuring the VLAN

If IP phones are connected to a switch, the switch notifies IP phones of the VLAN

information defined on the switch (if using LLDP). IP phones can then proceed with the

If in-line power switch/hub is provided, you don’t need to connect the phone to the power adapter. Make sure the switch/hub is PoE-compliant.

IP phones can also share the network with another network device such as a PC (personal computer). It is an optional connection.

Important! Do not unplug or remove the power while IP phones are updating firmware and configurations.

Getting Started

DHCP request for its network settings (if using DHCP).

Querying the DHCP (Dynamic Host Configuration Protocol) Server

IP phones are capable of querying a DHCP server. DHCP is enabled on IP phones by

default. The following network parameters can be obtained from the DHCP server

during initialization:

 IP Address

 Subnet Mask

 Gateway

 Primary DNS (Domain Name Server)

 Secondary DNS

You need to configure the network parameters of IP phones manually if any of them is

not supplied by the DHCP server. For more information on configuring network

parameters manually, refer to Configuring Network Parameters Manually on page 21.

Contacting the auto provisioning server

SIP-T4X IP phones support the FTP, TFTP, HTTP, and HTTPS protocols for auto provisioning

and are configured by default to use TFTP protocol. If IP phones are configured to obtain

configurations from the TFTP server, they will connect to the TFTP server and download

the configuration file(s) during startup. IP phones will be able to resolve and apply the

configurations written in the configuration file(s). If IP phones do not obtain the

configurations from the TFTP server, IP phones will use the configurations stored in the

flash memory.

Updating firmware

If the access URL of the firmware is defined in the configuration file, the IP phone will

download the firmware from the provisioning server. If the MD5 value of the

downloaded firmware file differs from that of the image stored in the flash memory, the

IP phone will perform a firmware update.

Downloading the resource files

In addition to configuration file(s), IP phones may require resource files before it can

deliver service. These resource files are optional, but if some particular features are

being deployed, these files are required.

The followings show examples of resource files:

 Language packs

 Ring tones

 Contact files

Administrator’s Guide for SIP-T4X IP Phones

After connected to the power and network, the IP phone begins the initializing process

by cycling through the following steps:

1. The power indicator LED illuminates.

2. The message “Initializing…Please wait” appears on the LCD screen when the IP

phone starts up.

3. The main LCD screen displays the following:

 Time and date

 Soft key labels

4. Press the OK key to check the IP phone status, the LCD screen displays the valid IP

address, MAC address, firmware version, etc.

If the IP phone has successfully passed through these steps, it starts up properly and is

ready for use.

You can use the following methods to set up and configure IP phones:

 Phone User Interface

 Web User Interface

 Configuration Files

The following sections describe how to configure IP phones using each method above.

An administrator or a user can configure and use IP phones via phone user interface.

Specific features access is restricted to the administrator. These specific features are

password protected by default. The default password is “admin“(case-sensitive). Not

all features are available on phone user interface.

An administrator or a user can configure IP phones via web user interface. The default

user name and password for the administrator to log into the web user interface are

both “admin” (case-sensitive). Almost all features are available for configuring via web

user interface. IP phones support both HTTP and HTTPS protocols for accessing the web

user interface. For more information, refer to Web Server Type on page 112.

Getting Started

You can deploy IP phones using configuration files. There are two configuration files

both of which are CFG formatted. We call them Common CFG file and MAC-Oriented

CFG file. A Common CFG file will be effectual for all IP phones of the same model.

However, a MAC-Oriented CFG file will only be effectual for a specific IP phone. The

Common CFG file has a fixed name for each IP phone model, while the MAC-Oriented

CFG file is named as the MAC address of IP phones. For example, if the MAC address of

a SIP-T46G IP phone is 001565113af5, the names of these two configuration files must be:

y000000000028.cfg and 001565113af5.cfg.

The name of the Common CFG file for each SIP-T4X IP phone model is:

 SIP-T46G: y000000000028.cfg

 SIP-T42G: y000000000029.cfg

 SIP-T41P: y000000000036.cfg

In order to deploy IP phones using configuration files (<y0000000000xx>.cfg and

<MAC>.cfg), you need to use a text-based editing application to edit the configuration

files, and store configuration files to a provisioning server. IP phones support

downloading configuration files using any of the following protocols: FTP, TFTP, HTTP and

HTTPS.

IP phones can obtain the address of the provisioning server during startup through one

of the following processes: Zero Touch, PnP, DHCP Options and Phone Flash. Then IP

phones download configuration files from the provisioning server, resolve and update

the configurations written in the configuration files. This entire process is called auto

provisioning. For more information on auto provisioning, refer to

Yealink_SIP-T2

Series_T19P_T4_Series_IP_Phones_Auto_Provisioning_Guide

When modifying parameters, learn the following:

 Parameters in configuration files override those stored in IP phones’ flash memory.

 The .cfg extension of the configuration files must be in lowercase.

 Each line in a configuration file must use the following format and adhere to the

following rules:

variable-name = value

- Associate only one value with one variable.

- Separate variable name and value with equal sign.

- Set only one variable per line.

- Put the variable and value on the same line, and do not break the line.

- Comment the variable on a separated line. Use the pound (#) delimiter to

distinguish the comments.

Administrator’s Guide for SIP-T4X IP Phones

IP phones can accept two sources of configuration data:

 Downloaded from configuration files

 Changed on the phone user interface or the web user interface

The latest value configured on the IP phone takes effect finally.

Icons associated with different features may appear on the LCD screen. The following

table provides a description for each icon on SIP-T4X IP phone models.

SIP-T46G

SIP-T42G/T41P

Description

Network is unavailable

Registered successfully

Registering

Hands-free speakerphone mode

Handset mode

Headset mode

Voice Mail /

Text Message

Auto Answer

Do Not Disturb

Call Forward

Call Hold

Call Mute

Ringer volume is 0

Getting Started

SIP-T46G

SIP-T42G/T41P

Description

Phone Lock

Multi-lingual lowercase letters input

mode

Multi-lingual uppercase letters input

mode

Alphanumeric input mode

Numeric input mode

Multi-lingual uppercase and

lowercase letters input mode

Received Calls

Placed Calls

Missed Calls

Recording box is full

A call cannot be recorded

Recording starts successfully

Recording cannot be started

Recording cannot be stopped

VPN is enabled

Bluetooth /

Bluetooth headset is both paired and

connected

Conference /

The default contact icon

The default caller photo

Administrator’s Guide for SIP-T4X IP Phones

This section describes how to configure basic network parameters for the IP phone.

Note

DHCP (Dynamic Host Configuration Protocol) is a network protocol used to dynamically

allocate network parameters to network hosts. The automatic allocation of network

parameters to hosts eases the administrative burden of maintaining an IP network. IP

phones comply with the DHCP specifications documented in RFC 2131. If DHCP is used,

IP phones connected to the network become operational without having to be manually

assigned IP addresses and additional network parameters. Static DNS address(es) can

be configured and used when DHCP is enabled.

DHCP Option

DHCP provides a framework for passing information to TCP/IP network devices. Network

and other control information are carried in tagged data items that are stored in the

options field of the DHCP message. The data items themselves are also called options.

DHCP can be initiated by simply connecting the IP phone with the network. IP phones

broadcast DISCOVER messages to request the network information carried in DHCP

options, and the DHCP server responds with the specific values in the corresponding

options.

The following table lists the common DHCP options supported by IP phones.

Parameter

DHCP Option

Description

Subnet Mask

Specify the client’s subnet mask.

Time Offset

Specify the offset of the client's subnet in

seconds from Coordinated Universal Time

(UTC).

Router

Specify a list of IP addresses for routers on the

client’s subnet.

Time Server

Specify a list of time servers available to the

client.

Domain Name

Server

Specify a list of domain name servers

available to the client.

Log Server

Specify a list of MIT-LCS UDP servers

This section mainly introduces IPv4 network parameters. For more information on IPv6, refer to IPv6 Support on page 187.

Getting Started

Parameter

DHCP Option

Description

available to the client.

Host Name

Specify the name of the client.

Domain Server

Specify the domain name that client should

use when resolving hostnames via DNS.

Broadcast

Address

Specify the broadcast address in use on the

client's subnet.

Network Time

Protocol

Servers

Specify a list of the NTP servers available to

the client by IP address.

Vendor-Specific

Information

Identify the vendor-specific information.

Vendor Class

Identifier

Identify the vendor type.

TFTP Server

Name

Identify a TFTP server when the 'sname' field

in the DHCP header has been used for DHCP

options.

Bootfile Name

Identify a bootfile when the 'file' field in the

DHCP header has been used for DHCP

options.

Procedure

DHCP can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure DHCP on the IP phone.

Configure static DNS address

when DHCP is used.

For more information, refer to

DHCP on page 248.

Local Web User Interface

Configure DHCP on the IP phone.

Configure static DNS address

when DHCP is used.

Navigate to:

http://<phoneIPAddress>/servlet

?p=network&q=load

Phone User Interface

Configure DHCP on the IP phone.

Administrator’s Guide for SIP-T4X IP Phones

To configure DHCP via web user interface:

1. Click on Network->Basic.

2. In the IPv4 Config block, mark the DHCP radio box.

3. Click Confirm to accept the change.

A dialog box pops up to prompt that settings will take effect after reboot.

4. Click OK to reboot the IP phone.

To configure static DNS address when DHCP is used via web user interface:

1. Click on Network->Basic.

2. In the IPv4 Config block, mark the DHCP radio box.

3. Mark the Static DNS radio box.

Getting Started

4. Enter the desired values in the Primary DNS and Secondary DNS fields.

5. Click Confirm to accept the change.

A dialog box pops up to prompt that settings will take effect after a reboot.

6. Click OK to reboot the IP phone.

To configure DHCP via phone user interface:

1. Press Menu->Advanced (password: admin) ->Network->WAN Port->IPv4.

2. Press or , or the Switch soft key to select the DHCP from the Type field.

3. Press the Save soft key to accept the change.

The IP phone reboots automatically to make settings effective after a period of time.

If DHCP is disabled or IP phones cannot obtain network parameters from the DHCP

server, you need to configure the network parameters manually. The following

parameters should be configured for IP phones to establish network connectivity:

 IP Address

 Subnet Mask

 Default Gateway

 Primary DNS

 Secondary DNS

Administrator’s Guide for SIP-T4X IP Phones

Procedure

Network parameters can be configured manually using the configuration files or locally.

Configuration File

<MAC>.cfg

Configure network parameters of

the IP phone manually.

For more information, refer to

Static Network Settings on page

249.

Local

Web User Interface

Configure network parameters of

the IP phone manually.

Navigate to:

http://<phoneIPAddress>/servlet

?p=network&q=load

Phone User Interface

Configure network parameters of

the IP phone manually.

To configure the IP address mode via web user interface:

1. Click on Network->Basic.

2. Select the desired value from the pull-down list of Mode (IPv4/IPv6).

3. Click Confirm to accept the change.

A dialog box pops up to prompt that settings will take effect after reboot.

4. Click OK to reboot the IP phone.

Getting Started

To configure a static IPv4 address via web user interface:

1. Click on Network->Basic.

2. In the IPv4 Config block, mark the Static IP Address radio box.

3. Enter the IP address, subnet mask, default gateway, primary DNS and secondary

DNS in the corresponding fields.

4. Click Confirm to accept the change.

A dialog box pops up to prompt that settings will take effect after reboot.

5. Click OK to reboot the IP phone.

To configure the IP address mode via phone user interface:

1. Press Menu->Advanced (password: admin) ->Network->WAN Port.

2. Press or to highlight the IP Address Mode field.

3. Press or to select IPv4 or IPv4&IPv6 from the IP Address Mode field.

4. Press the Save soft key to accept the change.

The IP phone reboots automatically to make settings effective after a period of time.

To configure a static IPv4 address via phone user interface:

1. Press Menu->Advanced (password: admin) ->Network->WAN Port->IPv4.

2. Press or , or the Switch soft key to select the Static IP from the Type field.

3. Enter the desired values in the IP Address, Subnet Mask, Gateway, Primary DNS

and Secondary DNS fields respectively.

4. Press the Save soft key to accept the change.

Administrator’s Guide for SIP-T4X IP Phones

The IP phone reboots automatically to make settings effective after a period of time.

Note

PPPoE (Point-to-Point Protocol over Ethernet) is a network protocol used by Internet

Service Providers (ISPs) to provide Digital Subscriber Line (DSL) high speed Internet

services. PPPoE allows an office or building-full of users to share a common DSL

connection to the Internet. PPPoE connection is supported by the Internet port of the IP

phone. Contact your ISP for the PPPoE user name and password. PPPoE is not applicable

to SIP-T42G and SIP-T41P IP phones.

Procedure

PPPoE can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure PPPoE on the IP phone.

For more information, refer to

PPPoE on page 252.

Local

Web User Interface

Configure PPPoE on the IP phone.

Navigate to:

http://<phoneIPAddress>/servlet

?p=network&q=load

Phone User Interface

Configure PPPoE on the IP phone.

To configure PPPoE via web user interface:

1. Click on Network->Basic.

2. In the IPv4 Config block, mark the PPPoE radio box.

Using the wrong network settings may result in inaccessibility of your phone and may also have an impact on your network performance. For more information on these parameters, contact your network administrator.

Getting Started

3. Enter the user name and password in the corresponding fields.

4. Click Confirm to accept the change.

A dialog box pops up to prompt that settings will take effect after reboot.

5. Click OK to reboot the IP phone.

To configure PPPoE via phone user interface:

1. Press Menu->Advanced (password: admin) ->Network->WAN Port->IPv4.

2. Press or , or the Switch soft key to select the PPPoE from the Type field.

3. Enter the user name and password in the corresponding fields.

4. Press the Save soft key to accept the change.

The IP phone reboots automatically to make settings effective after a period of time.

There are two Ethernet ports on the back of IP phones: Internet port and PC port. Three

optional methods of transmission configuration for SIP-T4X IP phone Internet or PC

Ethernet ports:

 Auto-negotiation

 Half-duplex

 Full-duplex

Administrator’s Guide for SIP-T4X IP Phones

Auto-negotiation is configured for both Internet and PC ports on the IP phone by default.

Auto-negotiation

Auto-negotiation means that all connected devices choose common transmission

parameters (e.g., speed and duplex mode) to transmit voice or data over Ethernet. This

process entails devices first sharing transmission capabilities and then selecting the

highest performance transmission mode supported by both. You can configure the

Internet port and PC port on IP phones to auto-negotiate during the transmission.

Half-duplex

Half-duplex transmission refers to transmitting voice or data in both directions, but in

one direction at a time; this means one device can send data on the line, but not

receive data simultaneously. You can configure the half-duplex transmission on both

Internet port and PC port for IP phones to transmit in 10Mbps, 100Mbps or 1000Mbps

(not applicable to SIP-T41P).

Full-duplex

Full-duplex transmission refers to transmitting voice or data in both directions at the

same time; this means one device can send data on the line while receiving data. You

can configure the full-duplex transmission on both Internet port and PC port for IP

phones to transmit in 10Mbps, 100Mbps or 1000Mbps (not applicable to SIP-T41P).

Getting Started

Procedure

The transmission method of Ethernet port can be configured using the configuration files

or locally.

Configuration File

<y0000000000xx>.cfg

Configure the transmission

method of Ethernet port.

For more information, refer to

Internet and PC Ports

Transmission Methods on page

253.

Local

Web User Interface

Configure the transmission

method of Ethernet port.

Navigate to:

http://<phoneIPAddress>/servlet

?p=network-adv&q=load

To configure the transmission method of Ethernet port via web user interface:

1. Click on Network->Advanced.

2. Select the desired value from the pull-down list of WAN Port Link.

3. Select the desired value from the pull-down list of PC Port Link.

4. Click Confirm to accept the change.

Administrator’s Guide for SIP-T4X IP Phones

Regular expression, often called a pattern, is an expression that specifies a set of strings.

A regular expression provides a concise and flexible means to “match” (specify and

recognize) strings of text, such as particular characters, words, or patterns of characters.

Regular expression is used by many text editors, utilities, and programming languages

to search and manipulate text based on patterns.

Regular expression can be used to define IP phone dial plan. Dial plan is a string of

characters that governs the way for IP phones to process the inputs received from the IP

phone’s keypads. IP phones support the following dial plan features:

 Replace Rule

 Dial-now

 Area Code

 Block Out

You need to know the following basic regular expression syntax when creating dial

plan:

The dot “.” can be used as a placeholder or multiple placeholders for

any string. Example:

“12.” would match “123”, “1234”, “12345”, “12abc”, etc.

The “x” can be used as a placeholder for any character. Example:

“12x” would match “121”, “122”, “123”, “12a”, etc.

The dash “-” can be used to match a range of characters within the

brackets. Example:

“[5-7]” would match the number “5”, ”6” or ”7”.

The comma “,” can be used as a separator within the bracket.

Example:

“[2,5,8]” would match the number ”2”, “5” or “8”.

[]

The square bracket "[]" can be used as a placeholder for a single

character which matches any of a set of characters. Example:

"91[5-7]1234" would match “9151234”, “9161234”, “9171234”.

()

The parenthesis "( )" can be used to group together patterns, for

instance, to logically combine two or more patterns. Example:

"([1-9])([2-7])3" would match “923”, “153”, “673”, etc.

The “$” followed by the sequence number of a parenthesis means

the characters placed in the parenthesis. The sequence number

stands for the corresponding parenthesis. Example:

A replace rule configuration, Prefix: "001(xxx)45(xx)", Replace:

Getting Started

"9001$145$2". When you dial out "0012354599" on your phone, the IP

phone will replace the number with "90012354599". “$1” means three

digits in the first parenthesis, that is, “235”. “$2” means two digits in

the second parenthesis, that is, “99”.

Replace rule is an alternative string that replaces the numbers entered by the user. IP

phones support up to 100 replace rules, which can be created either one by one or in

batch using a replace rule template. For more information on the replace rule template,

refer to Replace Rule Template on page 219.

Procedure

Replace rule can be created using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Create the replace rule for the IP

phone.

For more information, refer to Dial

Plan on page 254.

Local

Web User Interface

Create the replace rule for the IP

phone.

Navigate to:

http://<phoneIPAddress>/servlet

?p=settings-dialplan&q=load

To create a replace rule via web user interface:

1. Click on Settings->Dial Plan->Replace Rule.

2. Enter the string in the Prefix field.

3. Enter the string in the Replace field.

Administrator’s Guide for SIP-T4X IP Phones

4. Enter the desired line ID in the Account field or leave it blank.

If you leave the field blank or enter 0, the replace rule will apply to all accounts on

the IP phone.

5. Click Add to add the replace rule.

Dial-now is a string used to match the numbers entered by the user. When entered

numbers match the predefined dial-now rule, IP phones will automatically dial out the

numbers without pressing the send key. IP phones support up to 100 dial-now rules,

which can be created either one by one or in batch using a dial-now rule template. For

more information on the dial-now template, refer to Dial-now Template on page 220.

Delay Time for Dial-now Rule

IP phones will automatically dial out the entered number, which matches the dial-now

rule, after a specified period of time.

Procedure

Dial-now rule can be created using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Create the dial-now rule for the IP

phone.

For more information, refer to Dial

Plan on page 254.

Configure the delay time for the

dial-now rule.

For more information, refer to Dial

Getting Started

Plan on page 254.

Local

Web User Interface

Create the dial-now rule for the IP

phone.

Navigate to:

http://<phoneIPAddress>/servlet

?p=settings-dialnow&q=load

Configure the delay time for the

dial-now rule.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

To create a dial-now rule via web user interface:

1. Click on Settings->Dial Plan->Dial-now.

2. Enter the desired value in the Rule field.

3. Enter the desired line ID in the Account field or leave it blank.

If you leave the field blank or enter 0, the dial-now rule will apply to all accounts on

the IP phone.

4. Click Add to add the dial-now rule.

To configure the delay time for the dial-now rule via web user interface:

1. Click on Features->General Information.

Administrator’s Guide for SIP-T4X IP Phones

2. Enter the desired time within 1-14 (in seconds) in the Time-Out for Dial-Now Rule

field.

3. Click Confirm to accept the change.

Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate

geographical areas in one country. When the entered numbers match the predefined

area code rule, the IP phone will automatically add the area code before the numbers

when dialing out them. IP phones only support one area code rule.

Procedure

Area code rule can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Create the area code rule and

specify the maximum and

minimum lengths of the entered

numbers.

For more information, refer to Dial

Plan on page 254.

Local

Web User Interface

Create the area code rule and

specify the maximum and

minimum lengths of entered

numbers.

Navigate to:

Getting Started

http://<phoneIPAddress>/servlet

?p=settings-areacode&q=load

To configure an area code rule via web user interface:

1. Click on Settings->Dial Plan->Area Code.

2. Enter desired values in the Code, Min Length (1-15) and Max Length (1-15) fields.

3. Enter the desired line ID in the Account field or leave it blank.

If you leave the field blank or enter 0, the area code rule will apply to all accounts

on the IP phone.

4. Click Confirm to accept the change.

Block out rule prevents users from dialing out specific numbers. When the entered

numbers match the predefined block out rule, the LCD screen prompts “Forbidden

Number”. IP phones support up to 10 block out rules.

Procedure

Block out rule can be created using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Create the block out rule for the

IP phone.

For more information, refer to Dial

Plan on page 254.

Local

Web User Interface

Create the block out rule for the

desired line.

Navigate to:

http://<phoneIPAddress>/servlet

Administrator’s Guide for SIP-T4X IP Phones

?p=settings-blackout&q=load

To create a block out rule via web user interface:

1. Click on Settings->Dial Plan->Block Out.

2. Enter the desired value in the BlockOut Number field.

3. Enter the desired line ID in the Account field or leave it blank.

If you leave the field blank or enter 0, the block out rule will apply to all accounts on

the IP phone.

4. Click Confirm to add the block out rule.

Configuring Basic Features

This chapter provides information for making configuration changes for the following

basic features:

 Power Indicator LED

 Contrast

 Backlight

 User Password

 Administrator Password

 Phone Lock

 Time and Date

 Language

 Logo Customization

 Softkey Layout

 Key as Send

 Hotline

 Call Log

 Missed Call Log

 Local Directory

 Live Dialpad

 Call Waiting

 Auto Redial

 Auto Answer

 Call Completion

 Anonymous Call

 Anonymous Call Rejection

 Do Not Disturb

 Busy Tone Delay

 Return Code When Refuse

 Early Media

 180 Ring Workaround

 Use Outbound Proxy in Dialog

 SIP Session Timer

Administrator’s Guide for SIP-T4X IP Phones

 Session Timer

 Call Hold

 Call Forward

 Call Transfer

 Network Conference

 Transfer on Conference Hang Up

 Directed Call Pickup

 Group Call Pickup

 Dialog-Info Call Pickup

 Call Return

 Call Park

 Web Server Type

 Calling Line Identification Presentation

 Connected Line Identification Presentation

 DTMF

 Suppress DTMF Display

 Transfer via DTMF

 Intercom

Power indicator LED indicates power status and phone status. There are six

configuration options for power indicator LED:

Common Power Light On

Common Power Light On allows the power indicator LED to be turned on.

Ring Power Light Flash

Ring Power Light Flash allows the power indicator LED to flash when the IP phone

receives an incoming call. If this option is disabled, the status of the power indicator LED

is determined by the option “Common Power Light On”.

Voice/Text Mail Power Light Flash

Voice/Text Mail Power Light Flash allows the power indicator LED to flash when the IP

phone receives a voice mail or a text message. If this option is disabled, the status of

the power indicator LED is determined by the option “Common Power Light On”.

Mute Power Light Flash

Mute Power Light Flash allows the power indicator LED to flash when a call is mute. If

Configuring Basic Features

this option is disabled, the status of the power indicator LED is determined by the option

“Common Power Light On”.

Hold/Held Power Light Flash

Hold/Held Power Light Flash allows the power indicator LED to flash when a call is

placed on hold or is held. If this option is disabled, the status of the power indicator LED

is determined by the option “Common Power Light On”.

Talk/Dial Power Light On

Talk/Dial Power Light On allows the power indicator LED to be turned on when the IP

phone is busy. If this option is disabled, the status of the power indicator LED is

determined by the option “Common Power Light On”.

Procedure

Power indicator LED can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure the power indicator

LED.

For more information, refer to

Power Indicator LED on page 258.

Local

Web User Interface

Configure the power indicator

LED.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-powerled&q=load

To configure the power Indicator LED via web user interface:

1. Click on Features->Power LED.

2. Select the desired value from the pull-down list of Common Power Light On.

3. Select the desired value from the pull-down list of Ring Power Light Flash

4. Select the desired value from the pull-down list of Voice/Text Mail Power Light Flash.

5. Select the desired value from the pull-down list of Mute Power Light Flash.

6. Select the desired value from the pull-down list of Hold/Held Power Light Flash.

Administrator’s Guide for SIP-T4X IP Phones

7. Select the desired value from the pull-down list of Talk/Dial Power Light On.

8. Click Confirm to accept the change.

Contrast determines the readability of the texts displayed on the LCD screen. Adjusting

the contrast to a comfortable level can optimize the screen viewing experience. When

configured properly, contrast allows users to read the LCD’s display with minimal

eyestrain. For SIP-T46G IP phones, you can only configure the LCD’s contrast of the

connected EXP40. Make sure the expansion module has been connected to the IP

phone before adjustment.

Procedure

Contrast can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure the contrast of the LCD

screen.

For more information, refer to

Contrast on page 261.

Local

Phone User Interface

Configure the contrast of the LCD

screen.

Configuring Basic Features

To configure contrast via phone user interface (only applicable to EXP40 connected to

SIP-T46G IP phones):

1. Press Menu-> Basic->Display->Contrast.

2. Press or , or the Switch soft key to increase or decrease the intensity of

contrast.

The default contrast level is 6.

3. Press the Save soft key to accept the change.

Wallpaper is an image used as the background of the phone idle screen. Users can

select an image from IP phone’s built-in background or customize wallpaper from

personal pictures. To set the custom wallpaper as the phone background, you need to

upload the custom wallpaper to the IP phone in advance. The wallpaper is not

applicable to SIP-T42G and SIP-T41P IP phones.

The following table lists the supported wallpaper image format and resolution for

SIP-T46G IP phones:

Wallpaper Image Format

Resolution

Size

.jpg/.png/.bmp

<=480*272

<=5MB

Procedure

Wallpaper can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Specify the access URL of the

custom wallpaper.

For more information, refer to

Access URL of Wallpaper Image

on page 379.

Local

Web User Interface

Upload the custom wallpaper.

Change the wallpaper via web

user interface.

Navigate to:

http://<phoneIPAddress>/servlet

?p=settings-preference&q=load

Phone User Interface

Change the wallpaper via phone

user interface.

To upload custom wallpaper via web user interface:

1. Click on Settings->Preference.

Administrator’s Guide for SIP-T4X IP Phones

2. In the Upload Wallpaper (480*272) field, click Browse to locate the wallpaper

image from your local system.

3. Click Upload to upload the file.

4. Click Confirm to accept the change.

The custom wallpaper appears in the pull-down list of Wallpaper.

To change the wallpaper via web user interface:

1. Click on Settings->Preference.

2. Select the desired wallpaper from the pull-down list of Wallpaper.

3. Click Confirm to accept the change.

Configuring Basic Features

To change the wallpaper via phone user interface:

1. Press Menu->Basic->Display->Wallpaper.

2. Press or , or the Switch soft key to select the desired wallpaper.

3. Press the Save soft key to accept the change.

Backlight determines the brightness of the LCD screen display, allowing users to read

easily in dark environments. Backlight time specifies the delay time to turn off or dusky

the backlight when the IP phone is inactive. Backlight time is applicable to SIP-T4X IP

phones and EXP40 connected to SIP-T46G IP phones.

You can configure the backlight time as one of the following types:

 Always On: Backlight is turned on permanently.

 15, 30, 60, 120, 300, 600 or 1800: Backlight is turned off or turned dusky when the IP

phone is inactive after a preset period of time. It is automatically turned on if the

status of the IP phone changes or any key is pressed.

Backlight Active Level is used to adjust the backlight intensity of the LCD screen, and

Backlight Inactive Level is used to turn off or dusky the backlight after a period of

inactivity. Backlight Active Level is only applicable to SIP-T46G IP phones and the

connected EXP40. Backlight Inactive Level is only applicable to SIP-T46G IP phones.

Procedure

Backlight can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure the backlight of the

LCD screen.

For more information, refer to

Backlight on page 262.

Local

Web User Interface

Configure the backlight of the

LCD screen.

Navigate to:

http://<phoneIPAddress>/servlet

?p=settings-preference&q=load

Phone User Interface

Configure the backlight of the

LCD screen.

To configure the backlight via web user interface:

1. Click on Settings->Preference.

2. Select the desired value from the pull-down list of Backlight Inactive Level.

3. Select the desired value from the pull-down list of Backlight Active Level.

Administrator’s Guide for SIP-T4X IP Phones

4. Select the desired value from the pull-down list of Backlight Time (seconds).

5. Click Confirm to accept the change.

To configure the backlight via phone user interface:

1. Press Menu->Basic->Display-> Backlight.

2. Press or , or the Switch soft key to select the desired level from the

Backlight Active Level field.

3. Press or , or the Switch soft key to select the desired value from the

Backlight Inactive Level field.

4. Press or , or the Switch soft key to select the desired time from the

Backlight Time field.

5. Press the Save soft key to accept the change.

Some menu options are protected by two privilege levels, user and administrator, each

with its own password. When logging into the web user interface, you need to enter the

user name and password to access various menu options.

A user or an administrator can change the user password. The default user password is

“user”. For security reasons, the user or the administrator should change the default

user password as soon as possible.

Procedure

User password can be changed using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Change the user password of the

IP phone.

Configuring Basic Features

For more information, refer to

User Password on page 263.

Local

Web User Interface

Change the user password of the

IP phone.

Navigate to:

http://<phoneIPAddress>/servlet

?p=security&q=load

To change the user password via web user interface:

1. Click on Security->Password.

2. Select user from the pull-down list of User Type.

3. Enter a new password in the New Password and Confirm Password fields.

A new password should contain at least 6 characters, where at least one numeric

and one alphabetic characters. Valid characters are A-Z, a-z, 0-9,#,!,@,-,.,*,+ and $.

4. Click Confirm to accept the change.

Note

Advanced menu options are strictly used by administrators. Users can configure them

only if they have administrator privileges.

The administrator password can only be changed by an administrator. The default

administrator password is “admin”. For security reasons, the administrator should

change the default administrator password as soon as possible.

Procedure

Administrator password can be changed using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Change the administrator

password.

For more information, refer to

If logging into the web user interface of the phone with the user credential, user needs to enter the current user password in the Old Password field.

Administrator’s Guide for SIP-T4X IP Phones

Administrator Password on page

263.

Local

Web User Interface

Change the administrator

password.

Navigate to:

http://<phoneIPAddress>/servlet

?p=security&q=load

Phone User Interface

Change the administrator

password.

To change the administrator password via web user interface:

1. Click on Security->Password.

2. Select admin from the pull-down list of User Type.

3. Enter the current administrator password in the Old Password field.

4. Enter a new administrator password in the New Password and Confirm Password

fields.

A new password should contain at least 6 characters, where at least one numeric

and one alphabetic characters. Valid characters contain A-Z, a-z, 0-9,#,!,@,-,.,*,+

and $.

5. Click Confirm to accept the change.

To change the administrator password via phone user interface:

1. Press Menu->Advanced (password: admin) ->Set Password.

2. Enter the current administrator password in the Current Password field.

3. Enter a new administrator password in the New Password field and Confirm

Password field.

4. Press the Save soft key to accept the change.

Phone lock is used to lock the IP phone to prevent it from unauthorized use. Once the IP

phone is locked, a user must enter the password to unlock it. IP phones offer three types

Configuring Basic Features

of phone lock: Menu Key, Function Keys and All Keys. The IP phone will not be locked

immediately after the phone lock type is configured. One of the following steps is also

needed:

- Long press the pound key when the IP phone is idle.

- Press the keypad lock key (if configured) when the IP phone is idle.

In addition to the above steps, you can configure IP phones to automatically lock the

keypad after a period of time.

Procedure

Phone lock can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure the phone lock type.

Change the unlock password.

Configure the IP phone to

automatically lock the keypad

after a time interval.

For more information, refer to

Phone Lock on page 264.

Assign a keypad lock key.

For more information, refer to

Keypad Lock Key on page 386.

Local

Web User Interface

Configure the phone lock type.

Change the unlock password.

Configure the IP phone to

automatically lock the keypad

after a time interval.

Navigate to:

http://<phoneIPAddress>/servl

et?p=features-phonelock&q=lo

Assign a keypad lock key.

Navigate to:

http://<phoneIPAddress>/servl

et?p=dsskey&model=1&q=loa

d&linepage=1

Phone User Interface

Configure the phone lock type.

Assign a keypad lock key.

To configure phone lock via web user interface:

1. Click on Features->Phone Lock.

Administrator’s Guide for SIP-T4X IP Phones

2. Select the desired type from the pull-down list of Keypad Lock Enable.

3. Select the desired type from the pull-down list of Keypad Lock Type.

4. Enter unlock password (numeric characters) in the Phone Unlock PIN (0~15 Digit)

field.

5. Enter the desired time in the Phone Lock Time Out (0~3600s) field.

6. Click Confirm to accept the change.

To configure a keypad lock key via web user interface:

1. Click on DSSKey->Line Key.

2. In the desired DSS key field, select Keypad Lock from the pull-down list of Type.

3. Click Confirm to accept the change.

To configure phone lock type via phone user interface:

1. Press Menu->Advanced (password: admin) ->Phone Settings->Keypad Lock.

Configuring Basic Features

2. Press or , or the Switch soft key to select the desired value from the

Keypad Lock Enable field.

3. Press or , or the Switch soft key to select the desired type from the Keypad

Lock Type field.

4. Press the Save soft key to accept the change.

To configure a keypad lock key via phone user interface:

1. Press Menu->Call Features->DSS Keys.

2. Select the desired DSS key.

3. Press or , or the Switch soft key to select Key Event from the Type field.

4. Press or , or the Switch soft key to select Keypad Lock from the Key Event

field.

5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.

6. Press the Save soft key to accept the change.

IP phones maintain a local clock and calendar. Time and date are displayed on the idle

screen of the IP phone. Time and date are synced automatically from the NTP server by

default. The NTP server can be obtained by DHCP or configured manually.If IP phones

cannot obtain the time and date from the NTP server, you need to manually configure

them. The time and date display can use one of several different formats.

Time Zone

A time zone is a region on Earth that has a uniform standard time. It is convenient for

areas in close commercial or other communication to keep the same time. When

configuring IP phones to obtain the time and date from the NTP server, you must set the

time zone.

Daylight Saving Time

Daylight Saving Time (DST) is the practice of temporary advancing clocks during the

summertime so that evenings have more daylight and mornings have less. Typically,

clocks are adjusted forward one hour at the start of spring and backward in autumn.

Many countries have used the DST at various times, details vary by location. The DST

can be adjusted automatically from the time zone configuration. Typically, there is no

need to change this setting.

Administrator’s Guide for SIP-T4X IP Phones

The following table lists available methods for configuring time and date:

Option

Methods of Configuration

Time Zone

Configuration Files

Web User Interface

Phone User Interface

Time

Web User Interface

Phone User Interface

Time Format

Configuration Files

Web User Interface

Phone User Interface

Date

Web User Interface

Phone User Interface

Date Format

Configuration Files

Web User Interface

Phone User Interface

Daylight Saving Time

Configuration Files

Web User Interface

Procedure

Configuration changes can be performed using the configuration files or locally.

Configuration File

<MAC>.cfg

Configure NTP by DHCP priority

feature.

Configure the NTP server, time

zone and DST.

Configure the time and date

formats.

For more information, refer to

Time and Date on page 266.

Local

Web User Interface

Configure NTP by DHCP priority

feature.

Configure the NTP server, time

zone and DST.

Configure the time and date

manually.

Configure the time and date

formats.

Configuring Basic Features

Navigate to:

http://<phoneIPAddress>/servlet

?p=settings-datetime&q=load

Phone User Interface

Configure the NTP server and

time zone.

Configure the time and date

manually.

Configure the time and date

formats.

To configure NTP by DHCP priority feature via web user interface:

1. Click on Settings->Time & Date.

2. Select the desired value from the pull-down list of NTP By DHCP Priority.

3. Click Confirm to accept the change.

To configure the NTP server, time zone and DST via web user interface:

1. Click on Settings->Time & Date.

2. Select Disabled from the pull-down list of Manual Time.

3. Select the desired time zone from the pull-down list of Time Zone.

4. Enter the domain names or IP addresses in the Primary Server and Secondary

Server fields respectively.

5. Enter the desired time interval in the Synchronism (15~86400s) field.

6. Select the desired value from the pull-down list of Daylight Saving Time.

If you select Enabled, do one of the following:

Administrator’s Guide for SIP-T4X IP Phones

- Mark the DST By Date radio box in the Fixed Type field.

Enter the start time in the Start Date field.

Enter the end time in the End Date field.

- Mark the DST By Week radio box in the Fixed Type field.

Select the desired values from the pull-down lists of DST Start Month, DST Start

Day of Week, DST Start Day of Week Last in Month, DST Stop Month, DST Stop

Day of Week and DST Stop Day of Week Last in Month.

Enter the desired time in the Start Hour of Day field.

Enter the desired time in the End Hour of Day field.

Configuring Basic Features

7. Enter the desired offset time in the Offset (minutes) field.

8. Click Confirm to accept the change.

To configure the time and date manually via web user interface:

1. Click on Settings->Time & Date.

2. Select Enabled from the pull-down list of Manual Time.

3. Enter the time and date in the corresponding fields.

4. Click Confirm to accept the change.

To configure the time and date format via web user interface:

1. Click on Settings->Time & Date.

2. Select the desired value from the pull-down list of Time Format.

3. Select the desired value from the pull-down list of Date Format.

Administrator’s Guide for SIP-T4X IP Phones

4. Click Confirm to accept the change.

To configure the NTP server and time zone via phone user interface:

1. Press Menu->Basic->Time & Date->General->SNTP Settings.

2. Press or , or the Switch soft key to select the time zone that applies to your

area from the Time Zone field.

The default time zone is "+8 China(Beijing)".

3. Enter the domain names or IP addresses in the NTP Server 1 and NTP Server 2 fields

respectively.

4. Press or, or the Switch soft key to select Automatic from the Daylight

Saving field.

5. Press the Save soft key to accept the change.

To configure the time and date manually via phone user interface:

1. Press Menu->Basic->Time & Date->General->Manual Settings.

2. Enter the specific time and date.

3. Press the Save soft key to accept the change.

To configure the time and date formats via phone user interface:

1. Press Menu->Basic->Time & Date->Format.

2. Press or , or the Switch soft key to select the desired date format from the

Date Format field.

3. Press or , or the Switch soft key to select the desired time format (12 Hour

or 24 Hour) from the Time Format field.

4. Press the Save soft key to accept the change.

IP phones support multiple languages. Languages used on the phone user interface

and web user interface can be specified respectively as required.

The following table lists the languages supported by the phone user interface and the

web user interface respectively.

Phone User Interface

Web User Interface

English

Simplified Chinese (not

applicable to SIP-T42G/T41P)

Traditional Chinese (not

applicable to SIP-T42G/T41P)

French

English

Simplified Chinese (not

applicable to SIP-T42G/T41P)

Traditional Chinese (not

applicable to SIP-T42G/T41P)

French

Configuring Basic Features

Phone User Interface

Web User Interface

German

Italian

Polish

Portuguese

Spanish

Turkish

German

Italian

Turkish

Portuguese

Spanish

Not all of the supported languages are available for selection. Languages available for

selection depend on language packs currently loaded to IP phones. You can make

languages available for use on the phone user interface by loading language packs to

the IP phone. Language packs can only be loaded using the configuration files.

The following table lists available languages and the associated language packs:

Available

Language

Associated Language Pack

for SIP-T42G/T41P

Associated Language Pack

for SIP-T46G

English

lang+English.txt

Simplified Chinese

lang-Chinese_S.txt

Traditional Chinese

lang-Chinese_T.txt

German

lang-German.txt

French

lang-French.txt

Italian

lang-Italian.txt

Polish

lang-Polish.txt

lang-Portuguese.txt

Portuguese

lang-Portuguese.txt

lang-Polish.txt

Spanish

lang-Spanish.txt

Turkish

lang-Turkish.txt

To update translation of a built-in language, the file name of the language file cannot be

changed. For more information, refer to

Yealink_SIP-T2

Series_T19P_T4_Series_IP_Phones_Auto_Provisioning_Guide.

Procedure

Loading language pack can only be performed using the configuration files.

Configuration File

<y0000000000xx>.cfg

Specify the access URL of the

language pack.

Administrator’s Guide for SIP-T4X IP Phones

For more information, refer to

Language on page 272.

The default language used on the phone user interface is English. The default language

used on the web user interface depends on the language preferences in the browser (if

the language is not supported by the IP phone, the web user interface uses English). You

can specify the languages for the phone user interface and web user interface.

Procedure

Specify the language for the web user interface or the phone user interface using the

configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Specify the languages for the

phone user interface and the

web user interface.

For more information, refer to

Language on page 272.

Local

Web User Interface

Specify the language for the web

user interface.

Navigate to:

http://<phoneIPAddress>/servlet

?p=settings-preference&q=load

Phone User Interface

Specify the language for the

phone user interface.

To specify the language for the web user interface via web user interface:

1. Click on Settings->Preference.

Configuring Basic Features

2. Select the desired language from the pull-down list of Language.

3. Click Confirm to accept the change.

To specify the language for the phone user interface via phone user interface:

1. Press Menu->Basic->Language.

2. Press or to select the desired language.

3. Press the Save soft key to accept the change.

Logo customization allows unifying the IP phone appearance or displaying a custom

image on the idle screen such as a company logo, instead of the default system logo.

Logo is not applicable to SIP-T46G IP phones. The logo file format must be *.dob, and the

resolution of SIP-T42G/T41P IP phones is 192*64 graphic.

Note

Procedure

The logo shown on the idle screen can be configured using the configuration files or

locally.

Configuration File

<y0000000000xx>.cfg

Configure the logo shown on the

idle screen.

For more information, refer to

Logo Customization on page 273.

Before uploading your custom logo to IP phones, ensure the logo file is correctly formatted. For more information on customizing a logo file, refer to

Yealink_SIP-T2

Series_T19P_T4_Series_IP_Phones_Auto_Provisioning_Guide

Administrator’s Guide for SIP-T4X IP Phones

Local

Web User Interface

Configure the logo shown on the

idle screen.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

To configure a custom logo via web user interface:

1. Click on Features->General Information.

2. Select Custom logo from the pull-down list of Use Logo.

3. Click Browse to select the logo file from your local system.

4. Click Upload to upload the file.

5. Click Confirm to accept the change.

The custom logo screen and the idle screen are displayed alternately.

Softkey layout is used to customize the soft keys at the bottom of the LCD screen to best

meet users’ requirements. It can be configured based on call states. In addition to

specifying which soft keys to display, you can determine their display order. You can

create a template about the softkey layout for the different call states. For more

information on the softkey layout template, refer to Softkey Layout Template on page

221.

Configuring Basic Features

The following table lists the soft keys available for IP phones in different states:

Call State

Default Soft Key

Optional Soft Key

CallFailed

NewCall

Empty

Switch

Cancel

CallIn

Answer

Forward

Silence

Reject

Empty

Switch

Connecting

Empty

Cancel

Empty

Switch SemiAttendTrans

Transfer

Empty

Cancel

Empty

Switch

Dialing

Send

IME

Delete

Cancel

Empty

History

Switch

Line

Favorite

GPickup

DPickup

RingBack

Empty

Cancel

Empty

Switch

SemiAttendTransBack

Transfer

Empty

Cancel

Empty

Switch

CC Talking

Talk

Transfer

HOLD

Empty

MUTE

Administrator’s Guide for SIP-T4X IP Phones

Call State

Default Soft Key

Optional Soft Key

Conference

Cancel

SWAP

NewCall

Switch

Answer

Reject

Hold

Transfer

Resume

NewCall

Cancel

Empty

Switch

Answer

Reject

Held

Empty

Cancel

Empty

Switch

Answer

Reject

NewCall

PreTrans

Transfer

IME

Delete

Cancel

Empty

Directory

Switch

Send

Conferenced

Empty

Hold

Split

Cancel

Empty

Switch

Answer

Reject

Mute

Manager

Procedure

Softkey layout can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Specify the access URL of the

softkey layout template.

For more information, refer to

Access URL of Softkey Layout

Template on page 375.

Local

Web User Interface

Configure the softkey layout.

Navigate to:

http://<phoneIPAddress>/servlet

Configuring Basic Features

?p=settings-softkey&q=load

To configure softkey layout via web user interface:

1. Click on Settings->Softkey Layout.

2. Select the desired value from the pull-down list of Custom Softkey.

3. Select the desired state from the pull-down list of Call States.

4. Select the desired soft key from the Unselected Softkeys column and click .

The selected soft key appears in the Selected Softkeys column. If more than four

soft keys are selected, a More soft key appears on the LCD screen, and the

selected soft keys are displayed in two pages.

5. Repeat the step 4 to add more soft keys to the Selected Softkeys column.

6. Click to remove the soft key from the Selected Softkeys column.

7. Click or to adjust the display order of the soft key.

8. Click Confirm to accept the change.

Key as send allows assigning the pound key or star key as a send key. Send sound

allows the IP phone to play a key tone when a user presses the send key. Key tone

allows the IP phone to play a key tone when a user presses any key. Send sound works

only if Key tone is enabled.

Procedure

Key as send can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure a send key.

Configure a key tone and send

tone.

Administrator’s Guide for SIP-T4X IP Phones

For more information, refer to Key

as Send on page 274.

Local Web User Interface

Configure a send key.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

Configure a key tone and send

tone.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-audio&q=load

Phone User Interface

Configure the send key.

To configure a send key via web user interface:

1. Click on Features->General Information.

2. Select the desired value from the pull-down list of Key As Send.

3. Click Confirm to accept the change.

To configure a key tone and send tone via web user interface:

1. Click on Features->Audio.

2. Select the desired value from the pull-down list of Key Tone.

Configuring Basic Features

3. Select the desired value from the pull-down list of Send Sound.

4. Click Confirm to accept the change.

To configure a send key via phone user interface:

1. Press Menu->Call Features->Others->General.

2. Press or , or the Switch soft key to select # or * from the Key as Send field,

or select Disabled to disable this feature.

3. Press the Save soft key to accept the change.

Note

A hotline is a point-to-point communication link in which a call is automatically directed

to the preset hotline number. The IP phone automatically dials out the hotline number

using the first available line after a time interval when off-hook. IP phones only support

one hotline number.

Procedure

Hotline can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure the hotline number.

Specify the time (in seconds) the

IP phone waits to automatically

dial out the hotline number.

For more information, refer to

Send tone works only if key tone is enabled. Key tone is enabled by default.

Administrator’s Guide for SIP-T4X IP Phones

Hotline on page 276.

Local

Web User Interface

Configure the hotline number.

Specify the time (in seconds) the

IP phone waits to automatically

dial out the hotline number.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

Phone User Interface

Configure the hotline number.

Specify the time (in seconds) the

IP phone waits to automatically

dial out the hotline number.

To configure hotline via web user interface:

1. Click on Features->General Information.

2. Enter the hotline number in the Hotline Number field.

3. Enter the delay time in the Hotline Delay (0~10s) field.

4. Click Confirm to accept the change.

To configure hotline via phone user interface:

1. Press Menu->Call Features->Others->Hotline.

2. Enter the hotline number in the Number field.

3. Enter the delay time in the Hotline Delay 0-10(s) field.

4. Press the Save soft key to accept the change.

Configuring Basic Features

Call log contains call information such as remote party identification, time and date,

and call duration. IP phones maintain a local call log. Call log consists of four lists:

Missed calls, Placed calls, Received calls and Forwarded calls. Each call log list

supports up to 100 entries. To store call information, you must enable the save call log

feature in advance.

Procedure

Call log can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure the call log.

For more information, refer to Call

Log on page 277.

Local Web User Interface

Configure the call log.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

Phone User Interface

Configure the call log.

To configure the call log via web user interface:

1. Click on Features->General Information.

2. Select the desired value from the pull-down list of Save Call log.

Administrator’s Guide for SIP-T4X IP Phones

3. Click Confirm to accept the change.

To configure the call log via phone user interface:

1. Press Menu->Call Features->Others->General.

2. Press or , or the Switch soft key to select the desired value from the History

Record field.

3. Press the Save soft key to accept the change.

Missed call log allows IP phones to display the number of the missed calls with an

indicator icon on the idle screen, and to log the missed calls in the missed calls list when

IP phones miss calls. It is configurable on a per-line basis. Once the user accesses the

Missed calls list, the prompt message and an indicator icon on the idle screen are

cleared.

Procedure

Missed call log can be configured using the configuration files or locally.

Configuration File

<MAC>.cfg

Configure the missed call log

feature.

For more information, refer to

Missed Call Log on page 277.

Local

Web User Interface

Configure the missed call log

feature.

Navigate to:

http://<phoneIPAddress>/servlet

?p=account-basic&q=load&acc

To configure missed call log via web user interface:

1. Click on Account.

2. Select the desired account from the pull-down list of Account.

3. Click on Basic.

Configuring Basic Features

4. Select the desired value from the pull-down list of Missed Call Log.

5. Click Confirm to accept the change.

The IP phone maintains a local directory. The local directory can store up to 1000

contacts and 48 groups (including the default groups: Company, Family and Friend).

When adding a contact to the local directory, in addition to name and phone numbers,

you can also specify the account, ring tone and group for the contact. Contacts and

groups can be added either one by one or in batch using a contact file. For more

information on the contact file, refer to Local Contact File on page 223.

Procedure

Configuration changes can be performed using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Specify the access URL of the

local contact file.

For more information, refer to

Access URL of Local Contact File

on page 377.

Local

Web User Interface

Add a new group and a contact

to the local directory.

Navigate to:

http://<phoneIPAddress>/servlet

?p=contactsbasic&q=load&num

=1&group=

Phone User Interface

Add a new group and a contact

to the local directory.

Administrator’s Guide for SIP-T4X IP Phones

To add a new group to the local directory via web user interface:

1. Click on Directory->Local Directory.

2. In the Group Setting block, enter the new group name in the Group field.

3. Select the desired group ring tone from the pull-down list of Ring.

4. Click Add to add the new group.

To add a contact to the local directory via web user interface:

1. Click on Directory->Local Directory.

2. Enter the name and the office, mobile or other numbers in the corresponding fields.

3. Select the desired ring tone from the pull-down list of Ring Tone.

4. Select the desired group from the pull-down list of Group.

5. Select the desired account from the pull-down list of Account.

Configuring Basic Features

6. Select the desired photo from the pull-down list of Photo.

It is not applicable to SIP-T42G and SIP-T41P IP phones.

7. Click Add to add the contact.

To add a group to the local directory via phone user interface:

1. Press Menu->Directory->Local Group.

2. Press the Add Group soft key.

3. Enter the desired group name in the Group Name field.

4. Press or to select the desired group ring tone from the Ring Tones field.

5. Press the Save soft key to accept the change or the Back soft key to cancel.

To add a contact to the local directory via phone user interface:

1. Press Menu->Directory->Local Group.

2. Select the desired contact group and press the Enter soft key.

3. Press the Add soft key.

4. Enter the name and the office, mobile or other numbers in the corresponding fields.

5. Press or , or the Switch soft key to select the desired account from the

Account field.

If Auto is selected, the IP phone will use the first available account when placing

calls to the contact from the local directory.

6. Press or , or the Switch soft key to select the desired ring tone from the Ring

field.

7. Press or , or the Switch soft key to select the desired photo from the Photo

field.

8. Press the Save soft key to accept the change.

Administrator’s Guide for SIP-T4X IP Phones

Live dialpad allows IP phones to automatically dial out the entered phone number after

a specified period of time.

Procedure

Live dialpad can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure live dialpad.

For more information, refer to Live

Dialpad on page 278.

Local

Web User Interface

Configure live dialpad.

Navigate to:

http://<phoneIPAddress>/servlet

?p=settings-preference&q=load

To configure live dialpad via web user interface:

1. Click on Settings->Preference.

2. Select the desired value from the pull-down list of Live Dialpad.

3. Enter the desired delay time in the Inter Digit Time (1~14s) field.

4. Click Confirm to accept the change.

Call waiting allows IP phones to receive a new call when there is already an active call.

The new call is presented to the user visually on the LCD screen. Call waiting tone

Configuring Basic Features

allows the phone to play a short tone, to remind the user audibly of a new incoming call

during conversation. Call waiting tone works only if call waiting is enabled.

The call waiting on code and call waiting off code configured on IP phones are used to

activate/deactivate the server-side call waiting feature. They may vary on different

servers.

Procedure

Call waiting and call waiting tone can be configured using the configuration files or

locally.

Configuration File

<y0000000000xx>.cfg

Configure call waiting.

For more information, refer to Call

Waiting on page 278.

Local Web User Interface

Configure call waiting.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

Phone User Interface

Configure call waiting.

To configure call waiting via web user interface:

1. Click on Features->General Information.

2. Select the desired value from the pull-down list of Call Waiting.

3. (Optional.) Enter the call waiting on code in the Call Waiting On Code field.

4. (Optional.) Enter the call waiting off code in the Call Waiting Off Code field.

Administrator’s Guide for SIP-T4X IP Phones

5. Click Confirm to accept the change.

To configure the call waiting tone via web user interface:

1. Click on Features->Audio.

2. Select the desired value from the pull-down list of Call Waiting Tone.

3. Click Confirm to accept the change.

To configure call waiting and call waiting tone via phone user interface:

1. Press Menu->Call Features->Call Waiting.

2. Press or , or the Switch soft key to select the desired value from the Call

Waiting field.

3. Press or , or the Switch soft key to select the desired value from the Play

Tone field.

4. (Optional.) Enter the call waiting on code in the On Code field.

5. (Optional.) Enter the call waiting off code in the Off Code field.

6. Press the Save soft key to accept the change.

Auto redial allows IP phones to redial a busy number after the first attempt. Both the

number of attempts and waiting time between redials are configurable.

Configuring Basic Features

Procedure

Auto redial can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure auto redial.

For more information, refer to

Auto Redial on page 280.

Local Web User Interface

Configure auto redial.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

Phone User Interface

Configure auto redial.

To configure auto redial via web user interface:

1. Click on Features->General Information.

2. Select the desired value from the pull-down list of Auto Redial.

3. Enter the desired time interval (in seconds) in the Auto Redial Interval (1~300s)

field.

The default time interval is 10s.

4. Enter the desired times in the Auto Redial Times (1~300) field.

The default times are 10.

5. Click Confirm to accept the change.

To configure auto redial via phone user interface:

1. Press Menu->Call Features->Others->Auto Redial.

Administrator’s Guide for SIP-T4X IP Phones

2. Press or , or the Switch soft key to select the desired value from the Auto

Redial field.

3. Enter the desired time in the Redial Interval field.

4. Enter the desired times in the Redial Times field.

5. Press the Save soft key to accept the change.

Auto answer allows IP phones to automatically answer an incoming call. IP phones will

not automatically answer the incoming call during a call even if auto answer is enabled.

You can specify a period of delay time before the IP phone automatically answers

incoming calls. Auto answer is configurable on a per-line basis.

Procedure

Auto answer can be configured using the configuration files or locally.

Configuration File

<MAC>.cfg

Configure auto answer.

For more information, refer to

Auto Answer on page 281.

<y0000000000xx>.cfg

Specify a period of delay time for

auto answer.

For more information, refer to

Auto Answer on page 281.

Local Web User Interface

Configure auto answer.

Navigate to:

http://<phoneIPAddress>/servlet

?p=account-basic&q=load&acc

Specify a period of delay time for

auto answer.

http://<phoneIPAddress>/servlet

?p=features-general&q=load

Phone User Interface

Configure auto answer.

To configure auto answer via web user interface:

1. Click on Account.

2. Select the desired account from the pull-down list of Account.

3. Click on Basic.

Configuring Basic Features

4. Select the desired value from the pull-down list of Auto Answer.

5. Click Confirm to accept the change.

To configure a period of delay time for auto answer via web user interface:

1. Click on Features->General Information.

2. Enter the desired time (in seconds) in the Auto-Answer Delay (1~4s) field.

3. Click Confirm to accept the change.

To configure auto answer via phone user interface:

1. Press Menu->Call Features->Auto Answer.

2. Select the desired line and then press the Enter soft key.

Administrator’s Guide for SIP-T4X IP Phones

3. Press or , or the Switch soft key to select the desired value from the Auto

Answer field.

4. Press the Save soft key to accept the change.

Call completion allows users to monitor the busy party and establish a call when the

busy party becomes available to receive a call. Two factors commonly prevent a call

from connecting successfully:

 Callee does not answer

 Callee actively rejects the incoming call before answering

IP phones support call completion using the SUBSCRIBE/NOTIFY method, which is

specified in draft-poetzl-sipping-call-completion-00, to subscribe to the busy party and

receive notifications of their status changes. Call completion is not applicable to

SIP-T42G and SIP-T41P IP phones.

Procedure

Call completion can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure call completion.

For more information, refer to Call

Completion on page 282.

Local Web User Interface

Configure call completion.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

Phone User Interface

Configure call completion.

To configure call completion via web user interface:

1. Click on Features->General Information.

Configuring Basic Features

2. Select the desired value from the pull-down list of Call Completion.

3. Click Confirm to accept the change.

To configure call completion via phone user interface:

1. Press Menu->Call Features->Others->Call Completion.

2. Press or , or the Switch soft key to select the desired value from the Call

Completion field.

3. Press the Save soft key to accept the change.

Anonymous call allows the caller to conceal the identity information displayed on the

callee’s screen. The callee’s phone LCD screen prompts an incoming call from

anonymity.

Example of anonymous SIP header:

Via: SIP/2.0/UDP 10.2.8.183:5063;branch=z9hG4bK1535948896

From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=128043702

To: <sip:1011@10.2.1.199>

Call-ID: 1773251036@10.2.8.183

CSeq: 1 INVITE

Contact: <sip:1012@10.2.8.183:5063>

Content-Type: application/sdp

Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER,

PUBLISH, UPDATE, MESSAGE

Administrator’s Guide for SIP-T4X IP Phones

Max-Forwards: 70

User-Agent: Yealink SIP-T46G 28.72.0.1

Privacy: id

Supported: replaces

Allow-Events: talk,hold,conference,refer,check-sync

P-Preferred-Identity: <sip:1012@10.2.1.199>

Content-Length: 302

The anonymous call on code and anonymous call off code configured on IP phones are

used to activate/deactivate the server-side anonymous call feature. They may vary on

different servers. Send Anonymous Code allows IP phones to send anonymous call

on/off code to the server.

Procedure

Anonymous call can be configured using the configuration files or locally.

Configuration File

<MAC>.cfg

Configure anonymous call.

For more information, refer to

Anonymous Call on page 282.

Local Web User Interface

Configure anonymous call.

Navigate to:

http://<phoneIPAddress>/servlet

?p=account-basic&q=load&acc

Phone User Interface

Configure anonymous call.

To configure the anonymous call via web user interface:

1. Click on Account.

2. Select the desired account from the pull-down list of Account.

3. Click on Basic.

4. Select the desired value from the pull-down list of Local Anonymous.

5. Select the desired value from the pull-down list of Send Anonymous Code.

6. (Optional.) Enter the anonymous call on code in the On Code field.

Configuring Basic Features

7. (Optional.) Enter the anonymous call off code in the Off Code field.

8. Click Confirm to accept the change.

To configure the anonymous call via phone user interface:

1. Press Menu->Call Features->Anonymous.

2. Select the desired line and then press Enter soft key.

3. Press or , or the Switch soft key to select the desired value from the Local

Anonymous field.

4. (Optional.) Press or , or the Switch soft key to select the desired value

from the Send Anonymous Code field.

5. (Optional.) Enter the anonymous call on code in the On Code field.

6. (Optional.) Enter the anonymous call off code in the Off Code field.

7. Press the Save soft key to accept the change.

Anonymous call rejection allows IP phones to automatically reject incoming calls from

callers whose identity has been deliberately concealed. The anonymous caller’s LCD

screen presents “Anonymity Disallowed”.

The anonymous call rejection on code and anonymous call rejection off code

configured on IP phones are used to activate/deactivate the server-side anonymous call

rejection feature. They may vary on different servers.

Administrator’s Guide for SIP-T4X IP Phones

Procedure

Anonymous call rejection can be configured using the configuration files or locally.

Configuration File

<MAC>.cfg

Configure anonymous call

rejection.

For more information, refer to

Anonymous Call Rejection on

page 284.

Local

Web User Interface

Configure anonymous call

rejection.

Navigate to:

http://<phoneIPAddress>/servlet

?p=account-basic&q=load&acc

Phone User Interface

Configure anonymous call

rejection.

To configure anonymous call rejection via web user interface:

1. Click on Account.

2. Select the desired account from the pull-down list of Account.

3. Click on Basic.

4. Select the desired value from the pull-down list of Anonymous Call Rejection.

5. (Optional.) Enter the anonymous call rejection on code in the On Code field.

6. (Optional.) Enter the anonymous call rejection off code in the Off Code field.

7. Click Confirm to accept the change.

To configure anonymous call rejection via phone user interface:

1. Press Menu->Call Features->Anonymous.

Configuring Basic Features

2. Select the desired line and then press Enter soft key.

3. Press or , or the Switch soft key to select the desired value from the

Anonymous Rejection field.

4. (Optional.) Enter the anonymous call rejection on code in the On Code field.

5. (Optional.) Enter the anonymous call rejection off code in the Off Code field.

6. Press the Save soft key to accept the change.

Do Not Disturb (DND) allows IP phones to ignore incoming calls. DND can be configured

on a phone or per-line basis depending on the DND mode. Two DND modes:

 Phone (default): DND feature is effective for all accounts on the IP phone.

 Custom: DND feature can be configured for each or all accounts.

A user can activate or deactivate DND using the DND soft key or DND key. DND

activated on IP phones disables the local call forward settings. The DND configurations

on IP phones may be overridden by the server settings.

The DND on code and DND off code configured on IP phones are used to

activate/deactivate the server-side DND feature. They may vary on different servers.

Return Message When DND

This feature defines the return code and the reason of the SIP response message for the

rejected incoming call when DND is enabled on IP phones. The caller’s LCD screen

displays the received reason.

Procedure

DND can be configured using the configuration files or locally.

Configuration File

<MAC>.cfg

Configure DND in the custom

mode.

For more information, refer to Do

Not Disturb on page 286.

<y0000000000xx>.cfg

Assign a DND key.

For more information, refer to DND

Key on page 386.

Configure the DND mode.

Configure DND in the phone

mode.

Specify return code and reason of

the SIP response message.

For more information, refer to Do

Administrator’s Guide for SIP-T4X IP Phones

Not Disturb on page 286.

Local

Web User Interface

Assign a DND key.

Navigate to:

http://<phoneIPAddress>/servlet?

p=dsskey&model=1&q=load&line

page=1

Configure DND.

Navigate to:

http://<phoneIPAddress>/servlet?

p=features-forward&q=load

Specify return code and reason of

the SIP response message.

Navigate to:

http://<phoneIPAddress>/servlet?

p=features-general&q=load

Phone User Interface

Assign a DND key.

Configure DND.

To configure a DND key via web user interface:

1. Click on DSSKey->Line Key.

2. In the desired DSS key field, select DND from the pull-down list of Type.

3. Click Confirm to accept the change.

To configure the DND feature via web user interface:

1. Click on Features->Forward & DND.

2. In the DND block, mark the desired radio box in the Mode field.

a) If you mark the Phone radio box:

1) Mark the desired radio box in the DND Status field.

Configuring Basic Features

2) (Optional.) Enter the DND on code in the DND On Code field.

3) (Optional.) Enter the DND off code in the DND Off Code field.

b) If you mark the Custom radio box:

1) Select the desired account from the pull-down list of Account.

2) Mark the desired value in the DND Status field.

3) (Optional.) Enter the DND on code in the DND On Code field.

Administrator’s Guide for SIP-T4X IP Phones

4) (Optional.) Enter the DND off code in the DND Off Code field.

3. Click Confirm to accept the change.

To specify the return code via web user interface:

1. Click on Features->General Information.

2. Select the desired type from the pull-down list of Return Code When DND.

Configuring Basic Features

3. Click Confirm to accept the change.

To configure a DND key via phone user interface:

1. Press Menu->Call Features->DSS Keys.

2. Select the desired DSS key.

3. Press or , or the Switch soft key to select Key Event from the Type field.

4. Press or , or the Switch soft key to select DND from the Key Event field.

5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.

6. Press the Save soft key to accept the change.

To configure DND in the phone mode via phone user interface:

1. Press the DND soft key or the DND key when the IP phone is idle.

To configure DND in the custom mode for a specific account via phone user interface:

1. Press the DND soft key or the DND key when the IP phone is idle.

The LCD screen displays a list of the accounts registered on the IP phone.

2. Press or to select the desired account.

3. Press or to select On to activate DND.

4. Press the Save soft key to accept the change.

To configure DND in the custom mode for all accounts via phone user interface:

1. Press the DND soft key or the DND key when the IP phone is idle.

The LCD screen displays a list of the accounts registered on the IP phone.

2. Press the All On soft key to activate DND for all accounts.

3. Press the Save soft key to accept the change.

Busy tone is audible to the other party, indicating that the call connection has been

broken when one party releases a call. Busy tone delay can define a period of time

during which the busy tone is audible.

Procedure

Busy tone delay can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure the busy tone delay

feature.

For more information, refer to

Busy Tone Delay on page 279.

Local

Web User Interface

Configure the busy tone delay

feature.

Administrator’s Guide for SIP-T4X IP Phones

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

To configure busy tone delay via web user interface:

1． Click on Features->General Information. 2． Select the desired value from the pull-down list of Busy Tone Delay (Seconds).

3． Click Confirm to accept the change.

Return code when refuse defines the return code and reason of the SIP response

message for call rejection. The caller’s LCD screen displays the reason according to the

return code received. Available return codes and reasons are:

 404 (Not found)

 480 (Temporarily not available)

 486 (Busy here)

Procedure

Return code for call rejection can be configured using the configuration files or locally.

Configuration File

<y0000000000xx>.cfg

Configure the return code when

refusing a call.

For more information, refer to

Configuring Basic Features

Return Code When Refuse on

page 289.

Local

Web User Interface

Configure the return code when

refusing a call.

Navigate to:

http://<phoneIPAddress>/servlet

?p=features-general&q=load

To specify the return code when refusing a call via web user interface:

1. Click on Features->General Information.

2. Select the desired value from the pull-down list of Return Code When Refuse.

3. Click Confirm to accept the change.

Early media refers to media (e.g., audio and video) played to the caller before a SIP

call is actually established. Current implementation supports early media through the

183 message. When the caller receives a 183 message with SDP before the call is

established, a media channel is established. This channel is used to provide the early

media stream for the caller.

Yealink SIP-T4X, SIP-T42G, SIP-T41P Administration Manual

Specifications and Main Features

Frequently Asked Questions

User Manual