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(1) Warranty
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ii
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GNU GPL INFORMATION
Yealink IP phone firmware contains third-party software under the GNU General Public License (GPL). Yealink uses
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The original GPL license, source code of components licensed under GPL and used in Yealink products can be
Yealink administrator guide is intended for administrators who need to properly configure,
customize, manage, and troubleshoot the smart media phones rather than end-users. This guide
will help you understand the VoIP network and SIP components, and provides descriptions of all
available phone features.
This guide describes three methods for configuring IP phones: central provisioning, web user
interface and phone user interface. It will help you perform the following tasks:
Configure your IP phone on a provisioning server
Configure your phone’s features and functions via web/phone user interface
Introduction
Troubleshoot some common phone issues
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 IP networking and a prior
knowledge of IP telephony concepts are necessary.
The information detailed in this guide is applicable to firmware version 80 or higher. The
firmware format is like x.x.x.x.rom. The second x from left must be greater than or equal to 80
(e.g., the firmware version of SIP-T58V IP phone: 58.80.0.5.rom).
Chapters in This Guide
This administrator guide includes the following chapters:
Chapter 1, “Product Overview” describes the smart media phones and expansion modules.
Chapter 2, “Getting Started” describes how Yealink phones fit in your network and how to
install and connect IP phones, and also gives you an overview of IP phone’s initialization
process.
Chapter 3, “Setting Up Your System” describes some essential information on how to set
up your phone network and set up your phone with a provisioning server.
Chapter 4, “Configuring Basic Features” describes how to configure the basic features on IP
phones.
Chapter 5, “Configuring Advanced Features” describes how to configure the advanced
features on IP phones.
Chapter 6, “Configuring AudioFeatures” describes how to configure the audio features on
IP phones.
Chapter 7, “Configuring Video Features” describes how to configure the video features on
v
Administrator’s Guide for SIP-T5 Series Smart Media Phones
IP phones.
Chapter 8, “Configuring SecurityFeatures” describes how to configure the security features
on IP phones.
Chapter 9, “Troubleshooting” describes how to troubleshoot IP phones and provides some
common troubleshooting solutions.
Chapter 10, “Appendix” provides the glossary, time zones, trusted certificates, auto
provisioning flowchart, reference information about IP phones compliant with RFC 3261,
SIP call flows, some other function lists (e.g., DSS keys, reading icons) and index.
Related Documentations
The following related documents for SIP-T58V/A, SIP-T56A and CP960 IP phones are available:
Quick Start Guides, which describe how to assemble IP phones and configure the most
basic features available on IP phones.
User Guides, which describe how to configure and use the basic and advanced features
available on IP phones via phone user interface.
Auto Provisioning Guide, which describes how to provision IP phones using the
configuration files.
The purpose of
Auto Provisioning Guide
is to serve as a basic guidance for provisioning
Yealink IP phones with a provisioning server. If you are new to this process, it is helpful to
read this guide.
Description of Configuration Parameters in CFG Files, which describes all configuration
parameters in configuration files.
Note that Yealink administrator guide contains most of parameters. If you want to find out
more parameters not listed in this guide, please refer to
Parameters in CFG Files
y000000000000.boot template boot file.
<y0000000000xx>.cfg and <MAC>.cfg template configuration files.
IP Phones Deployment Guide for BroadSoft UC-One Environments, which describes how to
guide.
Description of Configuration
configure BroadSoft features on the BroadWorks web portal and IP phones.
IP Phone Features Integrated with BroadSoft UC-One User Guide, which describes how to
configure and use IP phone features integrated with BroadSoft UC-One on Yealink IP
phones.
vi
When the SIP server type is set to BroadSoft, please refer to these two guides to have a
better knowledge of configuring and using features integrated with Broadsoft UC-One.
For support or service, please contact your Yealink reseller or go to Yealink Technical Support
online: http://support.yealink.com/.
Conventions Used in Yealink Documentations
Convention
Description
Bold
Highlights the web/phone user interface items such as menus, menu
selections, soft keys, or directory names when they are involved in a
procedure or user action (e.g., Click on Settings->Upgrade.).
Also used to emphasize text (e.g., Important!).
Italics
Used to show the format of examples (e.g.,
http(s)://[IPv6 address]
),
or to show the title of a section in the reference documentations
available on the Yealink Technical Support Website (e.g.,
Triggering
the IP phone to Perform the Auto Provisioning
).
Blue Text
Used for cross references to other sections within this
documentation (e.g., refer to Ring Tones on page 601), for
hyperlinks to non-Yealink websites (e.g., RFC 3315) or for hyperlinks
to Yealink Technical Support website.
Blue Text in Italics
Used for hyperlinks to Yealink resources outside of this
documentation such as the Yealink documentations (e.g.,
Indicates that you must enter information specific to phone or
network. For example, when you see <MAC>, enter your phone’s
12-digit MAC address. If you see <phoneIPAddress>, enter your
phone’s IP address.
->
Indicates that you need to select an item from a menu. For
example, Settings->Basic indicates that you need to select Basic
from the Settings menu via phone user interface.
Note: By default, the Settings menu locates on the second idle
screen. You need to swipe left/right to see it. Or, you can also
swipe down from the top of the screen to enter the control center
Yealink documentations contain a few typographic conventions and writing conventions.
You need to know the following basic typographic conventions to distinguish types of in-text
information:
Introduction
You also need to know the following writing conventions to distinguish conditional information:
Reading the Configuration Parameter Tables
The feature descriptions discussed in this guide include two tables. One is a summary table of
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
provisioning methods that you can use to configure the features. The other is a table of details
of the configuration parameters that you configure to make the features work.
This brief section describes the conventions used in the summary table and configuration
parameter table. In order to read the tables and successfully perform configuration changes, an
understanding of these conventions is necessary.
Summary Table Format
The following summary table indicates three provisioning methods (central provisioning, web
user interface and phone user interface, refer to Provisioning Methods for more information)
you can use to configure a feature. Note that the types of provisioning methods available for
each feature will vary; not every feature uses all these three methods.
The central provisioning method requires you to configure parameters located in CFG format
configuration files that Yealink provides. For more information on configuration files, refer to
Configuration Files on page 116. As shown below, the table specifies the configuration file name
and the corresponding parameters. That is, the MAC.cfg file contains the
parameter, and the y0000000000xx.cfg file contains the
account.X.auto_answer
feature.auto_answer_delay
parameter.
The web user interface method requires you to configure features by navigating to the specified
link. This navigation URL can help you quickly locate the webpage where you can configure the
feature.
viii
Configuration Parameter Table Format
Sometimes you will see the words “Refer to the following content” in the Permitted Values or
The following configuration parameter table describes the parameter that you can configure to
make the feature (e.g., auto answer) work.
Introduction
Note
Default field. It means the permitted value or the default value of the parameter has the model
difference or there are many permitted values of the parameter, you can get more details from
the following Description field.
The word “None” in the Web User Interface or Phone User Interface field means this feature
cannot be configurable via web/phone user interface.
The above table also indicates three methods for configuring the feature.
Method 1: Central Provisioning
This table specifies the details of
the auto answer feature. This parameter is disabled by default. If you want to enable the auto
answer feature, open the MAC.cfg file and locate the parameter name
Set the parameter value to “1” to enable the auto answer feature or “0” to disable the auto
answer feature.
Note that some parameters described in this guide contain one or more variables (e.g., X or Y).
But the variables in the parameters described in the CFG file are all replaced with specific value
in the scope of variable. You may need to assign a value to the variable before you search and
locate the specific parameter in the CFG file.
account.X.auto_answer
parameter, which enables or disables
account.X.auto_answer
.
For example, if you want to enable the auto answer feature for account 1, you need to locate the
account.1.auto_answer
account.1.auto_answer = 1). If you want to enable the audio codec 1 for account 1, you can
in the MAC.cfg file and then configure it as required (e.g.,
ix
Administrator’s Guide for SIP-T5 Series Smart Media Phones
locate the
account.1.codec.1.enable = 1).
The following shows a segment of MAC.cfg file:
account.1.codec.1.enable
in the MAC.cfg file and configure it as required (e.g.,
Method 2: Web User Interface
As described in the chapter Summary Table Format, you can directly navigate to the specified
webpage to configure the feature. You can also first log into the web user interface, and then
locate the feature field according to the web path (e.g., Account->Basic->Auto Answer) to
configure it as required.
As shown in the following illustration:
x
Introduction
To successfully log into the web user interface, you may need to enter the user name (default:
admin) and password (default: admin). For more information, refer to Web User Interface on
page 113.
Method 3: Phone User Interface
You can configure features via phone user interface. Access to the desired feature according to
the phone path (e.g., Settings->Features->Auto Answer->Account X) and then configure it as
required.
As shown in the following illustration:
Recommended References
For more information on configuring and administering other Yealink products not included in
this guide, refer to product support page at Yealink Technical Support.
To access the latest Release Notes or other guides for Yealink IP phones, refer to the Document
Downloadpage for your phone at Yealink Technical Support.
If you want to find Request for Comments (RFC) documents, type
http://www.ietf.org/rfc/rfcNNNN.txt
browser.
This guide mainly takes the SIP-T58V IP phones as example for reference. For more details on
other IP phones, refer to
For other references, look for the hyperlink or web info throughout this administrator guide.
Yealink phone-specific user guide
(NNNN is the RFC number) into the location field of your
.
Understanding VoIP Principle and SIP Components
This section mainly describes the basic knowledge of VoIP principle and SIP components, which
will help you have a better understanding of the phone’s deployment scenarios.
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
VoIP Principle
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 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
attributes of an end-to-end call.
SIP provides capabilities to:
Determine the location of the target endpoint -- SIP supports address resolution, name
mapping, and call redirection.
Determine 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 media capabilities that can be supported by all
endpoints.
xii
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 Components
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 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.
Introduction
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
preferential to use this method when not using an application layer firewall. Application layer
firewalls like to know what applications are flowing through 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.
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
Summary of Changes
This section describes the changes to this guide for each release and guide version.
Changes for Release 80, Guide Version 80.12
Documentations of the newly released CP960 IP phones have been added.
The following sections are new for this version:
CSTA Control on page 450
Real-Time Transport Protocol (RTP) Ports on page 592
Transient Noise Suppressor (TNS) on page 651
Noise Barrier Suppressor (NBS) on page 652
Major updates have occurred to the following sections:
Physical Features of IP Phones on page 2
Key Features of IP Phones on page 4
Connecting the IP Phones on page 10
Power Saving on page 157
Bluetooth on page 164
Intercom on page 412
Customizing a Local Contact File on page 270
Auto Answer on page 292
Appendix C: Trusted Certificates on page 770
Changes for Release 80, Guide Version 80.11
Major update had occurred to the following section:
Door Phone on page 441
xiv
Table of Contents
Table of Contents
Introduction.......................................................................... v
About This Guide ................................................................................................................................................... v
Chapters in This Guide ......................................................................................................................................... v
Related Documentations ................................................................................................................................... vi
Conventions Used in Yealink Documentations .......................................................................................vii
Reading the Configuration Parameter Tables ..........................................................................................vii
Summary Table Format .......................................................................................................................... viii
Configuration Parameter Table Format .............................................................................................. ix
Recommended References ............................................................................................................................... xi
Understanding VoIP Principle and SIP Components ............................................................................. xi
VoIP Principle ............................................................................................................................................... xii
SIP Components ........................................................................................................................................ xiii
Summary of Changes ....................................................................................................................................... xiv
Changes for Release 80, Guide Version 80.12 .............................................................................. xiv
Changes for Release 80, Guide Version 80.11 .............................................................................. xiv
Table of Contents ............................................................... xv
SIP IP Phone Models .............................................................................................................................................1
Physical Features of IP Phones ................................................................................................................2
Key Features of IP Phones .........................................................................................................................4
Getting Started ..................................................................... 9
What IP Phones Need to Meet .........................................................................................................................9
Yealink IP Phones in a Network .......................................................................................................................9
Connecting the IP Phones ............................................................................................................................... 10
Inserting the Camera (only applicable to SIP-T58V/A IP phones) ........................................ 11
Attaching the Stand and the Optional Wall Mount Bracket (not applicable to CP960 IP phones)
Adjust the angle of touch screen (only applicable to SIP-T58V/A IP phones)................. 14
Connecting the Handset and Optional Headset (not applicable to CP960 IP phones)14
Connecting the Power and Network ................................................................................................. 15
Connecting the Optional USB Flash Drive....................................................................................... 18
Connecting the Wired Expansion MIC CPE90 (Only Applicable to CP960 IP Phones) . 19
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
Connecting the Optional PC using a Micro USB Cable (Only Applicable to CP960 IP Phones) 19
Connecting the Optional External Speaker (Only Applicable to CP960 IP Phones) ...... 20
Initialization Process Overview ...................................................................................................................... 20
Configuring Transmission Methods of the Internet Port and PC Port ................................ 40
Configuring PC Port Mode .................................................................................................................... 44
Web Server Type ........................................................................................................................................ 46
IPv6 Support ................................................................................................................................................ 66
Keeping User Personalized Settings after Auto Provisioning .............................................. 132
Configuring Basic Features ............................................ 143
Power Indicator LED ........................................................................................................................................ 145
Power Saving ..................................................................................................................................................... 157
Bluetooth ............................................................................................................................................................. 164
Display Method on Dialing .......................................................................................................................... 186
xvi
Table of Contents
Time and Date ................................................................................................................................................... 188
NTP Time Server ...................................................................................................................................... 189
Time and Date Settings ........................................................................................................................ 194
Daylight Saving Time (DST) ................................................................................................................ 197
Language ............................................................................................................................................................. 205
Loading Language Packs ..................................................................................................................... 206
Specifying the Language to Use....................................................................................................... 213
Key As Send ........................................................................................................................................................ 223
Dial Plan ............................................................................................................................................................... 227
Dial Plan using XML Template Files ................................................................................................ 227
Dial Plan using Digit Map String Rules .......................................................................................... 242
Off Hook Hot Line Dialing ............................................................................................................................ 258
Search Source List In Dialing ....................................................................................................................... 260
Customizing a Super Search Template File ................................................................................. 260
Save Call Log ...................................................................................................................................................... 263
Call List Show Number ................................................................................................................................... 265
Local Directory .................................................................................................................................................. 268
Customizing a Local Contact File ..................................................................................................... 268
Configuring Local Directory ............................................................................................................... 275
Live Dialpad ........................................................................................................................................................ 280
Auto Redial ......................................................................................................................................................... 290
Auto Answer ....................................................................................................................................................... 292
IP Direct Auto Answer .................................................................................................................................... 297
Allow IP Call ........................................................................................................................................................ 299
Accept SIP Trust Server Only ....................................................................................................................... 300
Do Not Disturb (DND) .................................................................................................................................... 313
Busy Tone Delay ............................................................................................................................................... 325
Return Code When Refuse ........................................................................................................................... 327
Early Media ......................................................................................................................................................... 329
180 Ring Workaround .................................................................................................................................... 329
Use Outbound Proxy in Dialog .................................................................................................................. 330
Call Hold .............................................................................................................................................................. 337
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
Music on Hold (MoH) ........................................................................................................................... 341
Call Transfer ........................................................................................................................................................ 363
Local Conference .............................................................................................................................................. 366
Transfer Mode via Dsskey............................................................................................................................. 372
Group Call Pickup ............................................................................................................................................ 382
Dialog Info Call Pickup ................................................................................................................................... 389
Recent Call In Dialing ..................................................................................................................................... 392
Call Number Filter ............................................................................................................................................ 397
Call Park ............................................................................................................................................................... 399
Calling Line Identification Presentation (CLIP) ..................................................................................... 403
Connected Line Identification Presentation (COLP) .......................................................................... 407
SIP Send MAC .................................................................................................................................................... 425
Reserve # in User Name ................................................................................................................................ 429
Unregister When Reboot .............................................................................................................................. 433
Reboot in Talking ............................................................................................................................................. 436
Answer By Hand ............................................................................................................................................... 438
Call Recording Using Soft Key .................................................................................................................... 439
Door Phone ........................................................................................................................................................ 441
Mobile Account ................................................................................................................................................. 445
CSTA Control ...................................................................................................................................................... 450
Configuring Advanced Features .................................... 453
Remote Phone Book ....................................................................................................................................... 453
Customizing Remote Phone Book Template File ...................................................................... 453
Visual Alert and Audio Alert for BLF Pickup ................................................................................ 477
BLF LED Mode .......................................................................................................................................... 479
Configuring a BLF Key........................................................................................................................... 482
Busy Lamp Field (BLF) List ............................................................................................................................ 486
Call Recording Using DSS Keys (Record and URL Record) ............................................................. 524
Hot Desking ........................................................................................................................................................ 531
Action URI ........................................................................................................................................................... 560
Configuring Trusted IP Address for Action URI ......................................................................... 564
Scenario A - Capturing the Current Screen of the Phone ..................................................... 566
Scenario B - Placing a Call via Web User Interface ................................................................... 568
Server Redundancy.......................................................................................................................................... 569
Server Domain Name Resolution ..................................................................................................... 581
Static DNS Cache.............................................................................................................................................. 584
Real-Time Transport Protocol (RTP) Ports ............................................................................................. 592
Configuring Audio Features ........................................... 601
Redial Tone ......................................................................................................................................................... 601
Ring Tones .......................................................................................................................................................... 602
Distinctive Ring Tones .................................................................................................................................... 607
Voice Mail Tone ................................................................................................................................................ 621
Ringer Device for Headset ........................................................................................................................... 622
Packetization Time (PTime) ................................................................................................................ 638
Opus Sample Rate .................................................................................................................................. 640
Transfer via DTMF ................................................................................................................................... 661
Play Local DTMF Tone .......................................................................................................................... 663
Configuring Video Features ........................................... 683
Video Settings ................................................................................................................................................... 683
Video Codecs ..................................................................................................................................................... 685
Configuring Security Features ....................................... 689
User and Administrator Passwords........................................................................................................... 689
Auto-Logout Time ........................................................................................................................................... 691
IP Address Issues..................................................................................................................................... 746
Time and Date Issues ............................................................................................................................ 747
Phone Book Issues ................................................................................................................................. 748
System Log Issues .................................................................................................................................. 752
Other Issues .............................................................................................................................................. 763
Appendix B: Time Zones ............................................................................................................................... 769
- Built in microphone array, 360 degree voice pickup (only applicable to CP960 IP
phones)
Video Features (only applicable to SIP-T58V/A IP phones)
- Video codec: H264HP, H264, VP8
- Image codec: JPEG, PNG, BMP
- Adaptive bandwidth adjustment
Network Features
- SIP v1 (RFC 2543), v2 (RFC 3261)
- NAT Traversal: STUN mode
- DTMF: INBAND, RFC 2833, SIP INFO
- Proxy mode and peer-to-peer SIP link mode
- IP assignment: Static/DHCP/PPPoE
- VLAN assignment: LLDP/Static/DHCP/CDP
- Bridge mode for PC port (not applicable to CP960 IP phones)
- HTTP/HTTPS server
- DNS client
- NAT/DHCP server
- IPv6 support
- Wi-Fi
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
5
Administrator’s Guide for SIP-T5 Series Smart Media Phones
- TR-069
Security
- HTTPS (server/client)
- SRTP (RFC 3711)
- 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
- 802.1X authentication
Expansion Modules
This section introduces EXP50 expansion modules. The expansion modules are consoles you can
connect to Yealink IP phones to add DSS keys, which can be used to assign predefined
functionalities for quickly accessing features. If you want to configure the expansion module
keys, you have to connect the expansion module(s) to the IP phone in advance.
Expansion modules enable you to handle large volume of calls on a regular basis and expand
the functional capability of your IP phone. For more information on how to connect and use the
expansion module, refer to
The following lists the available physical features of the currently supported expansion modules:
Yealink EXP50 User Guide.
EXP50
Physical Features:
- Rich visual experience with 4.3” 272 x 480 pixel color screen
- 20 physical keys each with a dual-color LED
6
- 3 physical page keys
- Support up to 3 modules daisy-chain
- Only one expansion module is powered by the host phone
- 1*Mini USB port and 1*USB2.0 port for data in and out
Product Overview
7
Administrator’s Guide for SIP-T5 Series Smart Media Phones
8
Getting Started
This chapter describes where Yealink IP phones fit in your network and provides basic
installation instructions of SIP-T58V/T58A/T56A/CP960 IP phones.
This chapter provides the following sections:
What IP Phones Need to Meet
Yealink IP Phones in a Network
Connecting the IP Phones
Initialization Process Overview
Verifying Startup
Getting Started
What IP Phones Need to Meet
In order to operate as SIP endpoints in your network successfully, IP phones must meet the
following requirements:
A working IP network is established.
VoIP gateways are configured for SIP.
The latest (or compatible) firmware of IP phones is available.
A call server is active and configured to receive and send SIP messages.
Yealink IP Phones in a Network
Yealink IP phones can connect physically through a Category 5E (CAT 5E) cable to a Ethernet
LAN, and send and receive all data using the same packet-based technology. They can also
connect to the wireless network.
Since the IP phone is a data terminal, digitized audio being just another type of data from its
perspective, the phone is capable of vastly more than traditional business phones. Moreover,
Yealink IP phones run the same protocols as your office personal computer, which means that
many innovative applications can be developed without resorting to specialized technology.
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
There are many ways to set up a phone network using Yealink IP phones. The following shows
an example of a network setup:
Connecting the IP Phones
This section introduces how to install SIP-T58V/T58A/T56A/CP960 IP phones with components
in packaging contents.
1. Insert the camera (only applicable to SIP-T58V/A IP phones)
2. Attach the stand and the optional wall mount bracket (not applicable to CP960 IP phones)
3. Adjust the angle of touch screen (only applicable to SIP-T58V/A IP phones)
4. Connect the handset and optional headset (not applicable to CP960 IP phones)
5. Connect the power and network
6. Connect the optional USB flash drive
7. Connect the wired expansion MIC CPE90 (only applicable to CP960 IP phones)
8. Connect the optional PC using a micro USB cable (only applicable to CP960 IP phones)
10
Getting Started
The optional accessories are not included in packaging contents. You need to purchase them
separately if required.
The camera is connected to the USB port on the top of the phone. And the IP phone only
supports the Yealink original USB camera CAM50. You should purchase it separately for SIP-T58A
smart media phone.
9. Connect the optional external speaker (only applicable to CP960 IP phones)
Note
Inserting the Camera (only applicable to SIP-T58V/A IP
phones)
To insert the camera:
For SIP-T58V/A:
Note
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
Attaching the Stand and the Optional Wall Mount Bracket (not
applicable to CP960 IP phones)
To attach the stand and the optional wall mount bracket:
For SIP-T58V/A:
Desk Mount Method
12
Wall Mount Method (Optional)
For SIP-T56A:
The reversible tab has a lip which allows the handset to stay on-hook when the IP phone is
Yealink Wall Mount Quick
Getting Started
Desk Mount Method
Wall Mount Method (Optional)
Note
mounted vertically.
For more information on how to mount the IP phone to a wall, refer to
Installation Guide for Yealink IP Phones.
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
Adjust the angle of touch screen (only applicable to
SIP-T58V/A IP phones)
To adjust the angle of touch screen:
For SIP-T58V/A:
Connecting the Handset and Optional Headset (not applicable
to CP960 IP phones)
To connect the handset and optional headset:
For SIP-T58V/A:
14
For SIP-T56A:
Connecting the Power and Network
AC Power (Optional)
To connect the AC power and network (not applicable to CP960 IP phones):
1) Connect the DC plug of the power adapter to the DC5V port on the IP phone and connect
the other end of the power adapter into an electrical power outlet.
Getting Started
2) Connect the included Ethernet cable between the Internet port on the IP phone and the
one on the wall or switch/hub device port.
For SIP-T58V/A:
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
You can also connect the IP phone to a wireless network according to your office environment.
The use of the third-party
For SIP-T56A:
Note
For more information, refer to
The IP phone should be used with Yealink original power adapter only.
power adapter may cause the damage to the phone.
Yealink phone-specific user guide
.
Power over Ethernet (PoE)
With the included Ethernet cable, SIP-T58V/T58A/T56A IP phones can be powered from a
PoE-compliant switch or hub. CP960 IP phones can only be powered from a PoE adapter.
To connect the PoE (for SIP-T58V/T58A/T56A IP phones):
1) Connect the Ethernet cable between the Internet port on the IP phone and an available port
on the in-line power switch/hub.
For SIP-T58V/A:
16
Getting Started
If in-line power switch/hub is provided, you don’t need to connect the phone to the power
a PC (personal computer). It is an optional connection. We recommend that you use the Ethernet
configurations.
For SIP-T56A:
To connect the PoE adapter (for CP960 IP phones):
1) Connect the Ethernet cable between the Internet port on the IP phone and Data & Power
Out port on the PoE adapter.
Note
2) Connect the Ethernet cable between the Data In port on the PoE adapter and the one on
the wall or switch/hub device port.
3) Connect the power plug of the PoE adapter into an electrical power outlet.
For CP960:
adapter. Make sure the switch/hub is PoE-compliant.
SIP-T58V/T58A/T56A IP phones can also share the network with another network device such as
cable provided by Yealink.
Important! Do not unplug or remove the power while the IP phone is updating firmware and
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
For SIP-T58V/T58A/T56A, the USB port (on the rear of the phone) can also be used to connect
wireless mic CPW90. For more information, refer to
Yealink CP960 User Guide
.
Connecting the Optional USB Flash Drive
To connect a USB flash drive:
1) Insert a USB flash drive into the USB port on the phone.
For SIP-T58V/A:
Note
For SIP-T56A:
For CP960:
color-screen expansion module EXP50 or USB headset. The IP phone officially supports certain
USB headset models. For more information, refer to
IP Phone
.
Tested headset list compatible with Yealink
For more information on how to use EXP50, refer to
information on how to use USB headset, refer to the documentation from the manufacturer.
18
For CP960, the USB port can also be used to connect wireless mic charging cradle to charge the
Yealink EXP50 User Guide
. For more
Getting Started
Connecting the Wired Expansion MIC CPE90 (Only Applicable
to CP960 IP Phones)
You can connect optional wired expansion MIC CPE90 to enhance the room coverage of the
conference phone.
To connect the wired expansion MIC CPE90:
1) Connect the free end of the optional CPE90 cable to one of the MIC ports on the IP phone.
Connecting the Optional PC using a Micro USB Cable (Only
Applicable to CP960 IP Phones)
You can connect a PC to listen to the PC audio using your CP960 IP phone.
To connect a PC:
1. Connect the micro USB port of the IP phone and the USB port of the PC using a micro USB
cable.
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
Connecting the Optional External Speaker (Only Applicable to
CP960 IP Phones)
To connect an optional external speaker:
1. Connect the 3.5mm audio-out port of the IP phone to the external speaker using a 3.5mm
jack cable.
Initialization Process Overview
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 IP phone to the network and to an electrical supply, the IP 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
information defined on the switch (if using LLDP or CDP). The IP phone can then proceed with
the DHCP request for its network settings (if using DHCP). For more information on VLAN, refer
to VLAN on page 53.
20
Querying the DHCP (Dynamic Host Configuration Protocol) Server
The IP phone is capable of querying a DHCP server. DHCP is enabled on the IP phone by default.
The following network parameters can be obtained from the DHCP server during initialization:
IP Address
Getting Started
Subnet Mask
Gateway
Primary DNS (Domain Name Server)
Secondary DNS
You need to configure network parameters of the IP phone 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 32.
Contacting the provisioning server
If the IP phone is configured to obtain configurations from the provisioning server, it will
connect to the provisioning server and download the configuration file(s) during startup. The IP
phone will be able to resolve and update configurations written in the configuration file(s). If the
IP phone does not obtain configurations from the provisioning server, the IP phone will use
configurations stored in the flash memory. For more information, refer to Setting Up Your
Phones with a Provisioning Server on page 110.
Updating firmware
If the access URL of firmware is defined in the configuration file, the IP phone will download
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.
You can manually upgrade firmware if the IP phone does not download firmware from the
provisioning server. For more information, refer to Upgrading Firmware on page 124.
Downloading the resource files
In addition to configuration file(s), the IP phone 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
For more information on resource files, refer to Resource Files on page 118.
Verifying Startup
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/mute indicator LED illuminates solid red.
2. The message “Welcome Initializing… Please wait” appears on the touch screen when the IP
phone starts up.
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
3.The main touch screen displays the following:
Time and date
Android keys (for SIP-T58V/T58A/T56A)
Pre-installed applications (for CP960)
4.Tap Settings->Status to check the IP phone status, the touch 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.
22
Setting Up Your System
This section describes essential information on how to set up your phone network and set up
your phones with a provisioning server. It also provides instructions on how to set up a
provisioning server, how to deploy Yealink IP phones from the provisioning server, how to
upgrade firmware, and how to keep user personalized settings after auto provisioning.
This chapter provides the following sections:
Setting Up Your Phone Network
SettingUpYourPhoneswithaProvisioningServer
Setting Up Your Phone Network
Setting Up Your System
Yealink IP phones operate on an Ethernet local area network (LAN) or wireless network. Local
area network design varies by organization and Yealink IP phones can be configured to
accommodate a number of network designs.
In order to get your IP phones running, you must perform basic network setup, such as IP
address and subnet mask configuration. You can configure the IPv4 or IPv6 network parameters
for the phone. You can also configure the appropriate security (VLAN and/or 802.1X
authentication) and Quality of Service (QoS) settings for the IP phone.
This chapter describes how to configure all the network parameters for IP phones, and it
provides the following sections:
DHCP
DHCPOption
ConfiguringNetworkParametersManually
PPPoE
ConfiguringTransmissionMethods of the Internet Port and PC Port
Configuring PC Port Mode
WebServerType
Wi-Fi
VLAN
IPv6Support
VPN
Configuring the IP Phone for Use with a Firewall or NAT
QualityofService(QoS)
802.1XAuthentication
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Administrator’s Guide for SIP-T5 Series Smart Media Phones
Central Provisioning
(Configuration File)
<MAC>.cfg
Configure DHCP on the IP phone.
Parameter:
static.network.internet_port.type
Web User Interface
Configure DHCP on the IP phone.
Navigate to:
http://<phoneIPAddress>/servlet?m=mod_dat
a&p=network&q=load
Phone User Interface
Configure DHCP on the IP phone.
Parameter
Permitted Values
Default
static.network.internet_port.type
0, 1 or 2
0
Description:
Configures the Internet port type for IPv4.
0-DHCP
1-PPPoE
2-Static IP Address
Note: It works only if the value of the parameter “static.network.ip_address_mode” is set to 0
(IPv4) or 2 (IPv4 & IPv6). If you change this parameter, the IP phone will reboot to make the