Yealink SIP-T2xP IP Phone Family Administrator Guide_V2.1

Copyright 2012 YEALINK NETWORK TECHNOLOGY

Copyright 2012 Yealink Network Technology CO., LTD. All rights reserved. No parts of this

publication may be reproduced or transmitted in any form or by any means, electronic or

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its consent to downloading and printing copies of the content provided in this file only for private

use and not for redistribution. No parts of this publication may be subject to alteration,

modification or commercial use. Yealink Network Technology CO., LTD. will not be liable for any

damages arising from use of an illegally modified or altered publication.

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 BUT ARE 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 interferences 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.

About This Guide

v

The Yealink SIP-T2xP IP Phone Family Administrator Guide is considered to be an

administration-level version, which is intended for administrators who need to properly

configure, customize, manage, and troubleshoot Yealink IP phone systems rather than

the end-users of the IP phones. It includes details on the functioning and configuration of

the IP phones.

Many of the features are described in this guide involving the network settings, which

could affect the phones performance in the network. For this reason, an understanding

of IP networking and prior knowledge of IP telephony concepts are recommended.

This guide covers the Yealink SIP-T28P, T26P, T22P and T20P IP phones. The following

related documents for the SIP-T2xP IP phones are available:

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

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

IP phones.

User Guides, which describe the basic and advanced features available on the IP

phones.

Yealink Auto Provisioning User Guide, which describes how to auto provision the IP

phones using the configuration files.

Yealink Configuration Conversion Tool User Guide, which describes how to encrypt

the configuration files using the Configuration Conversion Tool.

.cfg and .cfg template configuration files.

Yealink IP Phones Deployment Guide for BroadWorks Environments, which

describes how to configure the BroadSoft features on the BroadWorks web portal

and the IP phones.

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

Support at http://www.yealink.com/index.php/Faq/lists/classid/2.

The information detailed in this guide is applicable to the firmware version 70 or higher.

This administrator guide is divided into the following chapters:

Chapter 1, Product Overview describes the SIP components and SIP IP phones.

Chapter 2, Getting Started describes how to install and connect the Yealink SIP IP

Administrators Guide for SIP-T2xP IP Phones

vi

phones and the IP phone interface methods.

Chapter 3, Configuring Basic Features describes how to configure the basic

features on the IP phones.

Chapter 4, Configuring Advanced Features describes how to configure the

advanced features on the IP phones.

Chapter 5, Security Features describes the encryption information of the IP

phones.

Chapter 6, Upgrading the Firmware describes how to upgrade the firmware of

the IP phones.

Chapter 7, Resource Files describes the resource files that can be downloaded

by the IP phones.

Chapter 8, Troubleshooting describes how to troubleshoot the IP phones and

provides some common troubleshooting solutions.

Chapter 9, Appendix provides the glossary, reference information about the IP

phones comply with RFC 3261, SIP call flows and the sample configuration file.

The following sections are new for this version:

BLF Call Park on page 44

Web Server Type on page 46

Tones on page 52

Action URL on page 60

Action URI on page 63

Resource Files on page 83

Appendix B: Configuration Parameters on page 101

Appendix E: Sample Configuration File on page 227

Major updates have occurred to the following sections:

Creating Dial Plan on page 21

Phone Lock on page 27

Time and Date on page 28

Busy Lamp Field on page 57

Shared Call Appearance on page 58

About This Guide

vii

Major updates have occurred to the following sections:

Creating Dial Plan on page 21

Do Not Disturb (DND) on page 35

Call Forward on page 38

802.1X Authentication on page 70

Table of Contents

1

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

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

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

Changes from Previous Versions ................................................................................................... vi

Changes from Version 1.0 ....................................................................................................... vi

Changes from Version 2.0 ...................................................................................................... vii

Table of Contents ..................................................................... 1

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

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

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

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

Physical Features of the SIP-T2xP IP Phones ........................................................................... 4

Key Features of the SIP-T2xP IP Phones ................................................................................... 8

Getting Started ....................................................................... 11

Connecting the Phone ................................................................................................................... 11

Installing the SIP-T28P and SIP-T26P IP Phones...................................................................... 11

Installing the SIP-T22P and SIP-T20P IP Phones...................................................................... 13

Initialization Process Overview .................................................................................................... 15

Verifying Startup ............................................................................................................................ 17

Configuration Interfaces ............................................................................................................... 17

Phone User Interface.............................................................................................................. 17

Web User Interface ................................................................................................................ 17

Configuration Files.................................................................................................................. 18

Reading Icons ................................................................................................................................ 19

Configuring Network Parameters Manually ............................................................................... 20

Creating Dial Plan ......................................................................................................................... 21

Replace Rule ........................................................................................................................... 22

Dial-now .................................................................................................................................. 23

Area Code............................................................................................................................... 23

Block Out ................................................................................................................................. 24

Configuring Basic Features .................................................... 25

User Password ............................................................................................................................... 26


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Administrator Password ................................................................................................................ 26

Phone Lock ..................................................................................................................................... 27

Time and Date ............................................................................................................................... 28

Language ....................................................................................................................................... 30

Loading Language Packs ...................................................................................................... 31

Specifying the Language to Use........................................................................................... 31

Missed Call Log ............................................................................................................................. 32

Local Directory ............................................................................................................................... 33

Call Waiting .................................................................................................................................... 34

Auto Redial ..................................................................................................................................... 35

Do Not Disturb (DND) .................................................................................................................... 35

Call Hold ......................................................................................................................................... 37

Call Forward .................................................................................................................................. 38

Call Transfer ................................................................................................................................... 39

Centralized Conference ................................................................................................................ 40

Transfer on Conference Hang Up ................................................................................................ 41

Directed Pickup Key ...................................................................................................................... 41

Group Pickup Key .......................................................................................................................... 42

Call Park Key .................................................................................................................................. 44

BLF Call Park ................................................................................................................................... 44

Hotline ............................................................................................................................................ 45

Web Server Type............................................................................................................................ 46

Caller ID Presentation ................................................................................................................... 46

Callee ID Presentation .................................................................................................................. 47

DTMF ............................................................................................................................................... 48

Suppressing the Display of DTMF Digits ..................................................................................... 49

Configuring Advanced Features............................................ 51

Distinctive Ring Tones .................................................................................................................... 51

Tones ............................................................................................................................................... 52

Remote Phonebook ....................................................................................................................... 54

LDAP ................................................................................................................................................ 55

Busy Lamp Field ............................................................................................................................. 57

Shared Call Appearance ............................................................................................................. 58

As-Feature-Event ........................................................................................................................... 59

Action URL ...................................................................................................................................... 60

Action URI ....................................................................................................................................... 63

Server Redundancy ....................................................................................................................... 65

Network Address Translation ....................................................................................................... 67

SNMP .............................................................................................................................................. 69

802.1X Authentication ................................................................................................................... 70

TR-069 Device Management ........................................................................................................ 72

Security Features .................................................................... 75

Table of Contents

3

Transport Layer Security ................................................................................................................ 75

Encrypting Configuration Files ..................................................................................................... 78

Changing the AES Keys on the IP Phone .............................................................................. 79

Upgrading the Firmware ........................................................ 81

Resource Files ......................................................................... 83

Replace Rule Template ................................................................................................................. 83

Dial-now Template ......................................................................................................................... 84

Local Contact File .......................................................................................................................... 85

Remote XML Phonebook ............................................................................................................... 87

Specifying the Access URL of Resource Files .............................................................................. 88

Troubleshooting ...................................................................... 91

Troubleshooting Methods ............................................................................................................. 91

Viewing Log Files .................................................................................................................... 91

Capturing Packets .................................................................................................................. 92

Enabling the Watch Dog Feature .......................................................................................... 93

Getting Information from Status Indicators .......................................................................... 93

Analyzing Configuration Files ............................................................................................... 94

Troubleshooting Solutions ............................................................................................................. 94

Why is the phone LCD screen blank? ................................................................................... 94

Why cant the phone obtain the IP address? ....................................................................... 94

Why does the phone display No Service? ....................................................................... 95

Why cant the phone upgrade successfully? ....................................................................... 95

Why doesnt the phone display time and date correctly? ................................................. 95

Why do I get poor audio during a call? ............................................................................... 95

Why doesnt the phone apply the configuration? ............................................................... 96

How to solve the IP conflict problem? .................................................................................. 96

How to upgrade the phone firmware in the recovery mode? ........................................... 96

How to reset your phone to factory configurations? ........................................................... 96

Appendix ................................................................................ 99

Appendix A: Glossary ................................................................................................................... 99

Appendix B: Configuration Parameters .................................................................................... 101

Setting Parameters in Configuration Files .......................................................................... 101

Basic and Advanced Parameters ....................................................................................... 101

Security Feature Parameters ............................................................................................... 159

Upgrading the Firmware ..................................................................................................... 161

Resource Files ....................................................................................................................... 163

Troubleshooting .................................................................................................................... 165

Configuring DSS Key ............................................................................................................ 166


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Appendix C: SIP (Session Initiation Protocol) ............................................................................ 177

RFC and Internet Draft Support .......................................................................................... 178

SIP Request ............................................................................................................................ 179

SIP Header ............................................................................................................................ 180

SIP Responses ....................................................................................................................... 181

SIP Session Description Protocol (SDP) Usage .................................................................. 184

Appendix D: SIP Call Flows ........................................................................................................ 184

Successful Call Setup and Disconnect ............................................................................... 185

Unsuccessful Call SetupCalled User is Busy .................................................................. 188

Unsuccessful Call SetupCalled User Does Not Answer ................................................ 191

Successful Call Setup and Call Hold .................................................................................. 194

Successful Call Setup and Call Waiting ............................................................................. 197

Call Transfer without Consultation ...................................................................................... 202

Call Transfer with Consultation ............................................................................................ 207

Always Call Forward ............................................................................................................ 213

Busy Call Forward ................................................................................................................ 216

No Answer Call Forward ..................................................................................................... 219

Call Conference .................................................................................................................... 222

Appendix E: Sample Configuration File .................................................................................... 227

Index ......................................................................................233

Product Overview

1

This chapter contains the following information about the Yealink SIP-T2xP 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 implement.

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 worldwide by service providers and enterprises for both voice and

video services over IP networks.

SIP

SIP (Session Initiation Protocol) is the Internet Engineering Task Forces (IETFs) 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 end points. 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.


2

SIP provides the capabilities to:

Determine the location of the target end point -- SIP supports address resolution,

name mapping, and call redirection.

Determine the media capabilities of the target end point -- Via Session Description

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

end points. Conferences are established using only the media capabilities that can

be supported by all end points.

Determine the availability of the target end point -- A call cannot be completed

because the target end point is unavailable, SIP determines whether the called

party is already on the phone or did not answer in the allotted number of rings. It

then returns a message indicating why the target end point was unavailable.

Establish a session between the origin and target end point -- The call can be

completed, SIP establishes a session between the end points. SIP also supports

mid-call changes, such as the addition of another end point to the conference or

the changing of a media characteristic or codec.

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

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

between the transferee and a new end point (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 this 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. Using this

Product Overview

3

method may be the preferred measure when not using an application layer firewall,

application layer firewalls like to know what applications are flowing though which

ports and it is possible using content types of other applications other than the one you

are trying to let through which has been denied.

User agent server (UAS)

UAS is the Server that hosts the application responsible for receiving the SIP requests

from a UAC, and on reception 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-topeer.

Typically, a SIP end point 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-T2xP IP phone family. The Yealink SIP-T2xP IP phones are

end points 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 end points. The SIP-T2xP 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 that support SIP.

The SIP-T2xP 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 history and

directory access. The SIP-T2xP IP phones also support advanced functionalities,

including LDAP, Busy Lamp Field, Shared Call Appearance and Centralized Conference.

The following IP phone models are described:

SIP-T28P

SIP-T26P

SIP-T22P

SIP-T20P

The Yealink SIP-T2xP IP phones comply with the SIP standard (RFC 3261), and they can

only be used within a network that supports this type of phone.

For the Yealink SIP-T2xP IP phones to successfully operate as SIP endpoints in your

network, they must meet the following requirements:

A working IP network is established.

Routers are configured for VoIP.


4

VoIP gateways are configured for SIP.

The latest (or compatible) firmware of the SIP-T2xP IP phones is available.

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

This section lists the available physical features of the SIP-T2xP IP phones.

SIP-T28P

Physical Features:

- TI TITAN chipset and TI voice engine

- 320x160 graphic LCD with 4-level grayscales

- 6 VoIP accounts, Broadsoft/Avaya/Asterisk validated

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

- 48 keys including 16 programmable keys

- 1xRJ9 (4P4C) handset port

- 1xRJ9 (4P4C) headset port

- 2xRJ45 10/100M Ethernet ports

- 1XRJ12 (6P6C) expansion module port

- 19 LEDs: 1xpower, 6xline, 1xmessage, 1xheadset, 10xmemory

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

- Power over Ethernet (IEEE 802.3af)

- Multi-language display

Product Overview

5

SIP-T26P

Physical Features:


- 132x64 graphic LCD







- 1XRJ12 (6P6C) expansion module port

- 16 LEDs: 1xpower, 3xline, 1xmessage, 1xheadset, 10xmemory





6

SIP-T22P

Physical Features:


- 132x64 graphic LCD







- 5 LEDs: 1xpower, 3xline, 1xmessage



- Wall-mounted


Product Overview

7

SIP-T20P

Physical Features:


- 3-line LCD with an icon line and 2x15 characters lines







- 4 LEDs: 1xpower, 2xline, 1xmessage



- Wall-mounted



8

In addition to the physical features introduced above, the SIP-T2xP IP phones also

support the following key features when running the latest firmware:

Phone Features

- Call Options: emergency call, call waiting, dial plan, call hold, call mute,

3-way conference, speed dial.

- Basic Features: DND, phone lock, auto redial, call transfer, hotline, call pickup,

call forward, calling party identity.

- Advanced Features: BLF/BLF list, shared call appearance, distinctive ring

tones, remote phonebook, SNMP, LDAP, 802.1x.

Codecs and Voice Features

- Wideband codec: G.722

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

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

- Full-duplex speakerphone with AEC

Network Features

- SIP v1 (RFC2543), v2 (RFC3261)

- NAT Traversal: STUN mode

- DTMF: INBAND, RFC2833, SIP INFO

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

- IP assignment: Static/DHCP/PPPoE

- Bridge/Router mode

- TFTP/DHCP/PPPoE 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

- TR-069

Product Overview

9

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


10

Getting Started

11

This chapter introduces the initialization of the SIP-T2xP IP phones, the installing and

connecting process of the IP phones which you need to follow.

This chapter provides the following major sections:

Connecting the Phone

Initialization Process Overview

Verifying Startup

Configuration Interfaces

Reading Icons

Configuring Network Parameters Manually

Creating Dial Plan

This section introduces how to install SIP-T2xP IP phones with the components in the

packing list.

1. Attach the Stand

2. Connect the Handset and optional Headset

3. Connect the Network and Power

Note

1) Attaching the Stand:

A headset is not provided in the packing list.


12

2) Connecting the Handset and optional Headset:

3) Connecting the Network and Power:

You have two options for power and network connections. Select a suitable way

according to your actual situation.

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 the phone and

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

2. Connect the supplied Ethernet cable between the Internet port on the phone and

the Internet port in your network or switch/hub device port.

Power over Ethernet

Using a regular Ethernet cable, the SIP-T28P and SIP-T26P IP phones can be powered

from a PoE (IEEE 802.3af) compliant switch or hub.

Getting Started

13

To connect the PoE:

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

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

1) Attaching the Stand:

Desk-mounted Method

Wall-mounted Method


14

2) Connecting the Handset and optional Headset:

3) Connecting 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 the phone and

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

2. Connect the supplied Ethernet cable between the Internet port on the phone and

the Internet port in your network or switch/hub device port.

Power over Ethernet

Using a regular Ethernet cable, the SIP-T22P and SIP-T20P IP phones can be powered

from a PoE (IEEE 802.3af) compliant switch or hub.

Getting Started

15

To connect the PoE:

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

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

Note

The initialization process of the IP phone is responsible for network connectivity and

operation of the IP phone in your local network.

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

begins its initialization process.

During the initialization process, the following events take place:

Loading the ROM file

The ROM file resides in the flash memory of the IP phone. The IP phone comes from the

factory with a ROM file preloaded. During initialization, the IP phone runs a bootstrap

loader that loads and executes the ROM file.

Configuring the VLAN

If the IP phone is connected to a switch, the switch notifies the IP phone of the VLAN ID

defined on the switch. The IP phone can then proceed with the DHCP request for its

network settings (if using DHCP).

If in-line power is provided, you dont need to connect the AC adapter. Make sure the

Ethernet cable and switch/hub are PoE compliant.

The IP phone can also share the network with other network device such as a PC

(personal computer). It is an optional connection.

Important! Do not unplug or remove power while the IP phone is updating firmware and

configuration.


16

DHCP (Dynamic Host Configuration Protocol)

The IP phone is capable of querying a DHCP server. DHCP is enabled on the IP phone

by default. The following network settings 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 IP phone manually if any of the network parameters is not

supplied by the DHCP server. For more information on manually configuring network

parameters, refer to Configuring Network Parameters Manually on page 20.

Contacting the TFTP server

If the IP phone is using the TFTP server to obtain its configurations, there should be a

configuration file or files on the TFTP server that the phone will request and download.

The IP phone resolves and applies the configurations written in the configuration file or

files. In the configuration file or files, SIP parameters that are required by the IP phone to

operate in a SIP VoIP environment are defined. If the phone is not obtaining its

configurations via the TFTP server, the phone will use configurations that are stored in

the flash memory.

Verifying the firmware version

If the phone determines that the firmware version defined in a configuration file differs

from the image it has stored in the flash memory, it performs a firmware upgrade. When

performing a firmware upgrade, the phone downloads the firmware image from the

TFTP server, programs the image into the flash memory, and reboots.

Loading the resource files

The IP phone may require resource files that are used by some of the advanced

features, which are also defined in the configuration file or files. These resource files are

optional, but if the particular feature is being employed, these files are required.

Some examples of resource files include:

Language files

Ring tones

Directories

Getting Started

17

After connecting the power and network, the IP phone begins the startup process by

cycling through the following steps:

1. The power indicator LED is illuminated.

2. The message Initializing, Please wait appears as phone starts up.

3. The main LCD screen displays the following:

Time and Date

Soft key labels (not supported by T20P)

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

address, MAC address and 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 ways to setup and configure the IP phone:

Phone User Interface

Web User Interface

Configuration Files

The following sections describe how to configure the IP phone using each way.

The IP phone user interface provides an easy way to access features and functions for

using and configuring the IP phone. Access to specific features and functions are

restricted to the administrator. A user can configure a subset of these features and

functions. Advanced Settings are password protected by default. Enter the

administrator password to unlock. The default password is admin (case-sensitive).

Not all features are available for configuring via phone user interface.

An administrator can configure the IP phone using the web user interface. The default

administrators name and password for logging in the web user interface are admin

(case-sensitive). Almost all features are available to be configured using the web user

interface. The phone web user interface supports both HTTP and HTTPS protocols, you

can configure the web server type, refer to Web Server Type on page 46 for more

information.


18

An administrator can configure the IP phone using the configuration files. There are two

configuration files both of which are CFG format. 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 the specific IP

phone. The common CFG file has a fixed name for each phone model, while the

MAC-Oriented CFG file is named with the MAC address of the IP phone. For example, a

SIP-T22P IP phone whose MAC address is 001565113af8, the two configuration files must

be: y000000000005.cfg and 001565113af8.cfg.

The name of the Common CFG file for each SIP-T2xP IP phone model is:

SIP-T28P: y000000000000.cfg

SIP-T26P: y000000000004.cfg

SIP-T22P: y000000000005.cfg

SIP-T20P: y000000000007.cfg

In order to configure the IP phone using the configuration files (.cfg

and .cfg), you need to use a text-based editing application to edit the

configuration files, and store the configuration files to the root directory of a

configuration server. The IP phone supports downloading the configuration files using

any of the following protocols: FTP, TFTP, HTTP and HTTPS. You can configure the type of

configuration server.

The IP phone gets the address of the configuration server during startup through any of

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

phone downloads the configuration files from the configuration server, resolves and

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

Auto Provisioning. For more information on auto provisioning, refer to the document

Yealink Auto Provisioning User Guide.

When modifying parameters, remember the following:

Parameters in the configuration files override those stored in the 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.

Getting Started

19

- Use white space before or after a variable or value.

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

distinguish the comments.

The IP phone can accept two sources of configuration data:

Downloaded from the configuration files

Changed on the phone user interface or the web user interface

The latest values to be applied to the IP phone are the values that take effect.

When using or configuring different features on the IP phone, a variety of icons may

appear on the LCD screen. The following table lists and describes icons that you might

see while using different IP phone models.

T28P T26P T22P T20P Description

Network unavailable

/

Private line registers

successfully

/

Shared line registers

successfully

/ Registered fail

/ Registering

Hands-free speakerphone

mode

Handset mode

Headset mode

Voice Mail

/ Text Message

Auto Answer


20

T28P T26P T22P T20P Description

Do Not Disturb

Call Forward

/ Call Hold

Call Mute

/ Ringer volume is 0

Phone Lock

Received Calls

Dialed Calls

Missed Calls

By default, DHCP is enabled on the IP phone. The IP phone can automatically derive the

network settings from the DHCP server. If DHCP is disabled or the phone cannot obtain

network settings, you need to manually configure them. The following network

parameters must be configured for the IP phone to operate in an IP network:

IP Address

Subnet Mask

Default Gateway

Primary DNS

Secondary DNS

Procedure

Network settings can be configured using the configuration files or locally.

Configuration File .cfg

Configure the basic network

parameters of the phone.

For more information, refer to

Getting Started

21

Network Settings on page 101.

Local

Web User Interface



Navigate to:

http:///cgi-bin/

ConfigManApp.com?Id=2




For more information, refer to the

SIP-T2xP User Guide.

To configure the network settings via web user interface:

1. Click on Network.

2. 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 save the change.

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 dial plan for the IP phone. Dial plan is a string of characters that governs the way

for the IP phone processing the inputs received from the phone keypad. The IP phone

supports 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.

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


22

"12x" would match "121", "122", "123", "12a", etc.

-

Numeric ranges are allowed within the brackets: Digit - Digit.

Example:

[5-7] would match the number5, 6or 7.

[]

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", etc.

()

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: "9([5-7]) (.)", Replace: "5$2". When

you dial out "96123" on your phone, the phone will replace the

number as "5123" and then dial out. $2 means the characters in the

second parenthesis, that is, 123.

Replace rule is an alternated string of characters that replaces the numbers dialed by

the user. You can specify at most 20 replace rules for the IP phone. You can create

multiple replace rules using a replace rule template and specify which phones are to

use the replace rules. For more information on the replace rule template, refer to

Replace Rule Template on page 83.

Procedure

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


Create the replace rule for the

phone.

For more information, refer to Dial

Plan on page 103.

Local Web User Interface

Create the replace rule for the

phone.

Navigate to:

http:///cgi-bin/




Getting Started

23

Dial-now is a string of characters that used to match the numbers dialed by the user,

when the dialed numbers match the predefined dial-now rule, the IP phone will

automatically dial out the numbers without pressing the send key. You can specify at

most 20 dial-now rules for the phone. You can create multiple dial-now rules using the

dial-now template and specify which phones are to use. For more information on the

dial-now template, refer to Dial-now Template on page 84.

Procedure

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


Create the dial-now rule for the

phone.


Plan on page 104.


Create the dial-now rule for the

phone.

Navigate to:

http:///cgi-bin/




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

geographical areas in one country. You can configure only one area code for the IP

phone. When dialing out numbers, the IP phone will automatically add the area code

before the dialed numbers. You can also specify the area code for the desired line and

the fixed lengths of the dialed numbers.

Procedure

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


Configure the area code and

specify the maximum and

minimum lengths of the dialed

numbers.


Plan on page 104.


24


Configure the area code and

specify the maximum and

minimum lengths of the dialed

numbers.

Navigate to:

http:///cgi-bin/




You can block the IP phone to dial out some specific numbers by configuring block out

numbers. You can create at most 10 block out numbers. When dialing out the block out

number, the phone LCD screen prompts the message Forbidden Number.

Procedure

Block out number can be specified using the configuration files or locally.


Specify the block out number for

the phone.


Plan on page 106.


Specify the block out number for

the phone.

Navigate to:

http:///cgi-bin/




Configuring Basic Features

25

The IP phone has specific features you can configure to customize your phone. This

chapter provides information for making configuration changes for the following basic

features:

User Password

Administrator Password

Phone Lock

Time and Date

Language

Missed Call Log

Local Directory

Call Waiting

Auto Redial

Do Not Disturb (DND)

Call Hold

Call Forward

Call Transfer

Centralized Conference

Transfer on Conference Hang Up

Directed Pickup Key

Group Pickup Key

Call Park Key

BLF Call Park

Hotline

Web Server Type

Caller ID Presentation

Callee ID Presentation

DTMF

Suppressing the Display of DTMF Digits


26

Several setting menus are protected with two privilege levels, user and administrator,

each with its own password. When logging in the web user interface, the IP phone will

prompt for the username and password before granting access to various menu options.

A user or an administrator can change the user password. The IP phone supports

alphanumeric characters (- *.+ are included) only in passwords.

Procedure

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


Change the user password of the

phone.

For more information, refer to on

User Password page 107.


Change the user password of the

phone.

Navigate to:

http:///cgi-bin/


To change the user password via web user interface:

1. Click on Security.

2. Mark the user radio box.

3. Enter the current user password in the Current Password field.

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


Advanced menu options are restricted to administrator. You must log in as administrator

to configure them when using the web user interface. The administrator password can

be only changed by the administrator. The IP phone supports alphanumeric characters

(- *.+ are included) only in passwords.

Procedure

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


Change the administrator

password.


Administrator Password on page


27

107.


Change the administrator

password.

Navigate to:

http:///cgi-bin/


To change the administrator password via web user interface:

1. Click on Security.

2. Mark the admin radio box.

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

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


A user or administrator can lock the IP phone to prevent it from unauthorized use. Once

the IP phone is locked, the user or administrator needs to enter the password to unlock

the phone. The default unlock password is 123. You can change the unlock password

and configure the interval to automatically lock the phone. The phone supports numeric

characters only in unlock passwords. The maximum length of the password is 15

characters. You can configure a DSS key to be keypad lock key. After enabling the

phone lock feature, the phone is locked after a configured period of inactivity. You can

long press the pound key or press the keypad lock key on the idle screen to lock the

phone immediately. You are only allowed to dial out the emergency numbers when the

phone is locked. You can specify one or more emergency numbers for the IP phone.

There are four types of "locks" that can apply to the IP phone:

Menu Key: The Menu key is locked. You cannot access the menu of the IP phone

until unlocked.

Function Keys: The function keys are locked.

All Keys: All keys are locked except the Volume key. The IP phone can only dial out

emergency numbers and answer incoming calls by lifting the handset, pressing the

Speakerphone key, pressing the HEADSET key or pressing the OK key.

Lock&Answer: All keys are locked, except the Menu key, the HEADSET key and the

Volume key. The IP phone can automatically answer the incoming calls and place

the previous conversation (if one) on hold, but cannot end the call.


28

Procedure

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


Configure the type of phone

lock.

Change the unlock password

and configure the interval to

automatically lock the phone.

Configure the emergency

numbers.


Phone Lock on page 107.

Assign a keypad lock key.


Keypad Lock Key on page 171.

Local

Web User Interface


lock.

Change the unlock password

and configure the interval to

automatically lock the phone.

Configure the emergency

numbers.

Navigate to:

http:///cgi-bi

n/ConfigManApp.com?Id=5

Assign a keypad lock key.

Navigate to:

http:///cgi-bi




lock.


the SIP-T2xP User Guide.

The IP phone maintains a local clock and calendar. Time and date display on the idle

screen of the IP phone. In addition to configuring the IP phone to obtain the time and

date from the Simple Network Time Protocol (SNTP) server, you can also manually set

the time and date, the time and date format and the daylight saving time.


29

The following table lists the available methods for each feature:

Feature Methods of Configuration

Set Time Web User Interface


Set Time Format

Configuration Files

Web User Interface


Set Time Zone Configuration Files

Web User Interface

Set Date Web User Interface


Set Date Format

Configuration Files

Web User Interface


Set Daylight Saving Time Configuration Files

Web User Interface

Time Zone

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

areas in close commercial or other communication to keep the same time. You can set

time zone using the Time Zone option in the web user interface or using the

parameters in the configuration files.

Daylight Saving Time

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

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

clocks are adjusted forward one hour near the start of spring and are adjusted

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.

Usually there is no need to change this setting.

Procedure

DST, time and date can be configured using the configuration files or locally.


Configure the DST, time and date

manually.


Time and Date on page 109.


30

Local

Web User Interface

Configure the DST, time and date

manually.

Navigate to:

http:///cgi-bin/





Configure the time and date

manually.



The IP phone supports displaying multiple languages. The languages displaying on the

phone user interface and web user interface can be specified respectively as required.

The following tables list the languages supported by the phone user interface.

The phone user interface supports the following languages:

Available Language Associated Language File

English lang+English.txt

Chinese_S lang-Chinese_S.txt

Chinese_T lang-Chinese_T.txt

Deutsch lang-Deutsch.txt

French lang-French.txt

Italian lang-Italian.txt

Portuguese lang-Portuguese.txt

Polish lang-Polish.txt

Spanish lang-Spanish.txt

Turkish lang-Turkish.txt

Note

The Chinese_S and Chinese_T languages are not applicable to the SIP-T20P IP phone.


31

All supported languages may not be available for selection. The available selected

languages are dependent on the language packs currently on the IP phone. You can

make languages available to display on the phone user interface by loading language

files to the IP phone. The language files and file name must be the one listed in the

above tables. You can only load language packs to the IP phone using the configuration

files.

Note

Procedure

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


Specify the access URL of the

language file.


Language on page 113.

The default language displays on the phone user interface is English. Once the

language pack(s) have been loaded, you can specify which language to use. You need

to specify the languages for the phone user interface and web user interface

respectively.

Procedure

Specify the display language using the configuration files or locally.


Specify the languages for the

phone user interface and the

web user interface.


Language on page 113.


Specify the language for the web

user interface.

Navigate to:

http:///cgi-bin/


You can only load and delete the language packs of the IP phone using the configuration

files.

The default (English) language pack for the phone user interface cannot be deleted.


32




Specify the language for the

phone user interface.



The IP phone can display the number of missed calls and log the missed calls in the

Missed Calls list. This missed call log feature is configurable on a per-account basis.

When enabled, the prompt message " New Missed Call" along with an

indicator icon display on the idle screen of the IP phone. Once the user accesses the

Missed Calls list, the prompt " New Missed Call" and the indicator icon on the

idle screen are cleared. When disabled, there is no indicator displaying on the LCD

screen, the IP phone does not log the missed call in the Missed Calls list.

Procedure

Missed call log can be configured using the configuration files or locally.


Configure the missed call log

feature.


Missed Call Log on page 115.


Configure the missed call log.

Navigate to:

http:///cgi-bin/


To configure the missed call log via web user interface:

1. Click on Account.

2. Select the desired account from the pull-down list of Account.

3. Select the desired value from the pull-down list of Missed Call Log.



33

The built-in phone directory can store up to 300 contacts. You can manually store the

frequently used names and numbers to the local directory. You can add multiple

contacts by downloading a contact file to the phone. For more information on the

contact file, refer to Local Contact File on page 85.

Procedure

Downloading the contact file can be performed using the configuration files or locally.


Specify the access URL of the

local contact file.


Access URL of Local Contact File

on page 164.

Local

Web User Interface

Download the contact file or add

the contact to the phone.

Navigate to:

http:///cgi-bin/





Add the contact to the local

directory directly.




34

When you are in an active call, the call waiting feature notifies you of the new

incoming call, and presents the incoming call visually on the phone LCD screen. If the

call waiting feature is disabled, the new incoming call will be automatically rejected

by the IP phone with a busy message.

Note

Call Waiting Tone

You can also enable or disable the playing of a short Call Waiting Tone" when there is

an incoming call on your IP phone. This feature is enabled by default. If you have Call

Waiting Tone enabled, and a call comes into the IP phone when you are in an active call,

a tone is audible to notify you of that incoming call. The Call Waiting Tone feature works

only if the Call Waiting feature is enabled.

Procedure

Call waiting and call waiting tone can be configured using the configuration files or

locally.


Configure the call waiting and

call waiting tone features.

For more information, refer to Call

Waiting on page 115.


Configure the call waiting and

call waiting tone features.

Navigate to:

http:///cgi-bin/


To configure the call waiting via web user interface:

1. Click on Phone->Features->General Information >>.

2. Select the desired value from the pull-down list of Call Waiting.


To configure the call waiting tone via web user interface:

1. Click on Phone->Features->Audio Settings>>.

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


If the call waiting feature is disabled and the IP phone is in an active call, the Missed

Calls list does not get updated with details of incoming calls.


35

The auto redial feature allows the IP phone to redial the dialed out number

automatically when the dialed number is busy. You can set the times of auto redial and

the seconds to wait between redial attempts. The IP phone will retry as many times as

configured till the dialed number is no longer busy.

Procedure

Auto redial can be configured using the configuration files or locally.


Configure the auto redial feature

on the IP phone.


Auto Redial on page 116.


Configure the auto redial feature.

Navigate to:

http:///cgi-bin/


To configure the auto redial via web user interface:


2. Select Enabled from the pull-down list of Auto Redial.

3. Enter the desired value in the Auto Redial Interval field.

4. Enter the desired value in the Auto Redial Times field.


The DND feature allows the IP phone to reject the incoming calls on a specific line or on

all lines. The incoming calls are logged as missed calls while the DND is enabled. You

can enable the DND on the phone or on the server. You can enable/disable the DND

feature for the phone, or you can customize the DND feature for each account or all

accounts. The following describes the DND key modes:

Phone (default): When the DND key mode is phone, it means the DND feature

applies to all accounts.

Custom: When the DND key mode is custom, it means that you can configure the

DND feature for each account or all accounts.

The DND enabled on the phone disables the local Call Forward settings. The DND

enabled on the IP phone may be overridden by the server settings. You can configure a

DSS key to be DND key. Press the DND key on the IP phone to toggle the DND mode.


36

Return Message When DND

When the DND mode is activated on the IP phone, any user calls the accounts

registered on the IP phone will be rejected automatically and receives a busy tone and

a message. The return message displays on the LCD screen of the callers phone. You

can configure the type of the return message.

Procedure

DND features can be configured using the configuration files or locally.


Specify the type of return

message.


Do Not Disturb on page 117.

Assign a DND key.

For more information, refer toDND

Key on page 171.

Specify the DND key mode.


Do Not Disturb on page 117.

Local

Web User Interface

Specify the type of the return

message.

Navigate to:

http:///cgi-bin/


Assign a DND key.

Navigate to:

http:///cgi-bin/




Specify the DND key mode.

Navigate to:

http:///cgi-bin/



Assign a DND key.



To configure the type of return message via web user interface:



37

2. Select the desired type from the pull-down list of Return Code When DND.


To configure the DND key mode via web user interface:

1. Click on Phone->Features->DND>>.

2. Select the desired mode in the DND Key Mode field.


Call Hold Tone

The call hold feature allows you to place an active call on hold while you want to initiate

or answer a second call. The line LED flashes green and the IP phone plays a warning

tone at regular intervals to remind you that there still has a call on hold. You can set the

interval for the IP phone to play hold tone. When the play hold tone feature is disabled,

the IP phone will not play a warning tone when there is a call on hold.

Call Hold Method

The IP phone supports two SIP call hold methods. One is according to RFC 2543, setting

c=0.0.0.0, the other is according to RFC 3261, setting c=IP address. The IP phone uses

RFC 3261 to request the remote party to stop sending media by default.

Procedure

Call hold can be configured using the configuration files or locally.


Configure the call hold tone and

call hold tone delay features.

Specify whether to use RFC 2543

(c=0.0.0.0) when outgoing hold

signaling.


Hold on page 117.


Configure the call hold tone and

call hold tone delay features.

Specify whether to use RFC 2543

(c=0.0.0.0) when outgoing hold

signaling.

Navigate to:

http:///cgi-bin/



38

To configure the call hold method via web user interface:


2. Select the desired value from the pull-down list of RFC 2543 Hold.


To configure the call hold tone and call hold tone delay features via web user interface:

1. Click on Phone->Features->Audio Settings>>.

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

3. Enter the desired time in the Play Hold Tone Delay field.


The call forward feature allows the IP phone to forward incoming calls to another party.

You can manually forward incoming calls to a predefined or a random number while the

phone is in the ringing state. You can also enable the call forward feature to statically

forward incoming calls to a predefined number. You can configure three types of call

forward:

Always Forward -- Forward the incoming calls immediately.

Busy Forward -- Forward the incoming call when the called user is busy.

No Answer Forward -- Forward the incoming call after a period of ring time.

You can configure the phone to support the call forward feature for the phone system, or

for each account. The following describes the call forward key modes:

Phone: Call forward in phone mode means that the call forward feature is effective

for the phone system.

Custom: Call forward in custom mode means that you can configure the call

forward feature for each account or all accounts.

You can set on code and off code on the phone to inform the server to sync the settings

of call forward configured on the IP phone. The on code and off code may vary in

different servers.

Procedure

Call forward can be configured using the configuration files or locally.


Specify the Call Forward key

mode.


Call Forward on page 118.

Configure the call forward


39

feature in phone mode.




feature in custom mode.



Local

Web User Interface


feature.

Navigate to:

http:///cgi-bi






feature.



The call transfer feature allows user to transfer an existing call to another party. The IP

phone offers three types of transfer:

Blind Transfer -- Transfer a call directly to another party without consulting.

Semi-attended Transfer -- Transfer a call after hearing the ring-back tone.

Attended Transfer -- Transfer a call with prior consulting.

Normally, the transfer is completed by pressing the Transfer key. You can configure the IP

phone to complete the blind transfer and attended transfer through on-hook. This

feature is enabled by default.

When performing the semi-attended transfer, you can configure the phone whether to

display the prompt 1 New Missed Call(s) on the LCD screen of the destination partys

phone. This feature is disabled by default.

Procedure

Call transfer can be configured using the configuration files or locally.


Configure the phone to complete

the blind or attended transfer

through on-hook.

Configure the semi-attended


40

transfer feature.


Transfer on page 123.


Configure the phone to complete

the blind or attended transfer

through on-hook.

Configure the semi-attended

transfer feature.

Navigate to:

http:///cgi-bin/


To configure the call transfer feature via web user interface:

1. Click on Phone->Features->Transfer Settings>>.

2. Select the desired values from the pull-down lists of Semi-Attended Transfer, Blind

Transfer on Hook and Attended Transfer on Hook.


You can initiate a local conference with the remote parties by using the phones local

audio processing resources. There is no dependency on network signaling for local

conference. The IP phone also supports centralized conference for which external

resources are used such as a conference bridge. The centralized conferences depend

on support from the SIP server.

Procedure

Centralized conference can be configured using the configuration files or locally.


Configure the type and URI of the

centralized conference.


Centralized Conference on page

128.


Configure the type and URI of the

centralized conference.

Navigate to:

http:///cgi-bin/





41

All parties release from the call when the conference initiator drops from the conference

call. You can enable the Transfer on Conference Hang up feature on the initiators

phone, the conference initiator will transfer the call when dropping from the conference

call. This feature is only applicable to the local conference.

Procedure

Transfer on conference hang up feature can be configured using the configuration files

or locally.


Configure the transfer on

conference hang up feature.


Transfer on Conference Hang Up

on page 128.


Configure the transfer on

conference hang up feature.

Navigate to:

http:///cgi-bin/


To configure the Transfer on Conference Hang up feature via web user interface:

1. Click on Phone->Features->Transfer Settings>>.

2. Select the desired value from the pull-down list of Transfer on Conference Hang

Up.


The directed pickup feature allows the user to pick up ringing calls of the specific

extension. When assigning a directed pickup key on the IP phone, specify the extension

you want to intercept. You can configure the IP phone to display the DPickup soft key in

the dialing interface. Then you can pick up the incoming calls of the monitored

extension using the DPickup soft key or the directed pickup key. When configuring the

DPickup soft key, you can configure the directed pickup code on a phone or per-account

basis. The settings on a per-account basis take precedence over the settings on a

phone basis. If the monitored extension receives multiple incoming calls, the IP phone

picks up the first incoming call. The directed pickup feature depends on support from

the SIP server.


42

Procedure

Directed pickup can be configured using the configuration files or locally.


Assign a directed pickup key.


Directed Pickup Key on page

172.

Configure the directed pickup

feature on a phone basis.


Directed

Yealink SIP-T2xP IP Phone Family Administrator Guide_V2.1

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