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Copyright © 2013 YEALINK NETWORK TECHNOLOGY
Copyright © 2013 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 mechanical, photocopying, recording, or otherwise, for any purpose, without the express written permission of Yealink Network Technology CO., LTD. Under the law, reproducing includes translating into another language or format.
When this publication is made available on media, Yealink Network Technology CO., LTD. gives its consent to downloading and printing copies of the content provided in this file only for private use but 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 AND PRESENTED
WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL
RESPONSIBILITY FOR THEIR APPLICATION OF PRODUCTS.
YEALINK NETWORK TECHNOLOGY CO., LTD. MAKES NO WARRANTY OF ANY KIND WITH
REGARD TO THIS GUIDE, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. Yealink Network Technology
CO., LTD. shall not be liable for errors contained herein nor for incidental or consequential damages in connection with the furnishing, performance, or use of this guide.
Hereby, Yealink Network Technology CO ., LTD. declares that this phone is in conformity with the essential requirements and other relevant provisions of the CE, FCC.
This device is marked with the CE mark in compliance with EC Directives 2006/95/EC and 2004/108/EC.
This device is compliant with Part 15 of the FCC Rules. Operation is subject to the following two conditions:
1. This device may not cause harmful interference, and
2. This device must accept any interference received, including interference that may cause undesired operation.
Note: This device is tested and complies with the limits for a Class B digital device, pursuant to Part 15 of the
FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:
1. Reorient or relocate the receiving antenna.
2. Increase the separation between the equipment and receiver.
3. Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
4. Consult the dealer or an experience radio/TV technician for help.
To avoid the potential effects on the environment and human health as a result of the presence of hazardous substances in electrical and electronic equipment, end users of electrical and electronic equipment should understand the meaning of the crossed-out wheeled bin symbol. Do not dispose of WEEE as unsorted municipal waste and have to collect such WEEE separately.
We are striving to improve our documentation quality and we appreciate your feedback. Email your opinions and comments to [email protected]
.
Yealink IP phone firmware contains third-party software under the GNU General Public License (GPL).
Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms and conditions of the license.
The original GPL license, source code of components licensed under GPL and used in Yealink products can be downloaded from Yealink web site: http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293 .
About This Guide
This guide is intended for administrators who need to properly configure, customize, manage, and troubleshoot the IP phone system rather than end-users. It provides details on the functionality and configuration of IP phones.
Many of the features described in this guide involve network settings, which could affect the IP phone’s performance in the network. So an understanding of IP networking and a prior knowledge of IP telephony concepts are necessary.
This guide covers SIP-T28P, SIP-T26P, SIP-T22P, SIP-T21P, SIP-T20P and SIP-T19P IP phones. The following related documents are available:
Quick Installation Guides, which describe how to assemble IP phones.
Quick Reference Guides, which describe the most basic features available on IP phones.
User Guides, which describe the basic and advanced features available on IP phones.
Auto Provisioning Guide, which describes how to provision IP phones using the configuration files.
<y0000000000xx>.cfg and <MAC>.cfg template configuration files.
IP Phones Deployment Guide for BroadSoft UC-One Environments, which describes how to configure BroadSoft features on the BroadWorks web portal and IP phones.
For support or service, please contact your Yealink reseller or go to Yealink Technical
Support online: http://www.yealink.com/Support.aspx
.
The information detailed in this guide is applicable to firmware version 72 or higher. The firmware format is like x.x.x.x.rom. The second x from left must be greater than or equal to 72 (e.g., the firmware version of SIP-T28P IP phone: 2.72.0.1.rom). This administrator guide includes 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 IP phones and
the configuration methods.
Chapter 3, “ Configuring Basic Features ” describes how to configure the basic
features on IP phones. v
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Chapter 4, “ Configuring Advanced Features ” describes how to configure the
advanced features on IP phones.
Chapter 5, “ Configuring Audio Features ” describes how to configure the audio
features on IP phones.
Chapter 6, “ Configuring Security Features ” describes how to configure the security
features on IP phones.
Chapter 7, “ Upgrading Firmware ” describes how to upgrade firmware of IP
phones.
Chapter 8, “ Resource Files ” describes the resource files that can be downloaded
by IP phones.
Chapter 9, “ Troubleshooting ” describes how to troubleshoot IP phones and
provides some common troubleshooting solutions.
Chapter 10, “ Appendix ” provides the glossary, reference information about IP
phones compliant with RFC 3261, SIP call flows and the sample configuration files.
This section describes the changes to this guide for each release and guide version. vi
The following section is new for this version:
Power Indicator LED on page 40
Major updates have occurred to the following sections:
Transport Layer Security on page 211
About This Guide
Upgrading Firmware on page 225
Documentations of the newly released SIP-T19P and SIP-T21P IP phones have also been added.
Major updates have occurred to the following sections:
on page
Major updates have occurred to the following sections:
on page
Distinctive Ring Tones on page 131
on page
Transport Layer Security on page 211
Secure Real-Time Transport Protocol
on page
Encrypting Configuration Files on page 219
Local Contact File on page 233
on page
Major updates have occurred to the following section:
Appendix B: Time Zones on page 257
Major updates have occurred to the following section:
Configuring DSS Key on page 399
vii
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
The following sections are new for this version:
on page
TR-069 Device Management on page 194
Major updates have occurred to the following sections:
Configuring Network Parameters Manually on page 24
on page
Directed Call Pickup on page 107
Distinctive Ring Tones on page 131
Automatic Call Distribution on page 149
Transport Layer Security on page 211
Local Contact File on page 233
viii
The following sections are new for this version:
Configuring Network Parameters Manually on page 24
on page
Anonymous Call Rejection on page 82
About This Guide
Return Code When Refuse on page 89
180 Ring Workaround on page 90
Use Outbound Proxy in Dialog on page 92
Automatic Call Distribution on page 149
Message Waiting Indicator on page 151
Quality of Service on page 185
Configuring Audio Features on page 199
Secure Real-Time Transport Protocol on page 217
on page
Phone user interface for each feature
Major updates have occurred to the following sections:
on page
Transport Layer Security on page 211
Encrypting Configuration Files on page 219
Web user interface for each feature
The following sections are new for this version:
Dialog Info Call Pickup on page 113
ix
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Appendix C: Configuration Parameters
on page
Appendix F: Sample Configuration File
on page
Major updates have occurred to the following sections:
on page
x
Table of Contents
About This Guide ...................................................................... v
Table of Contents .................................................................... xi
Product Overview ..................................................................... 1
Getting Started ....................................................................... 13
xi
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Configuring Basic Features .................................................... 39
xii
Table of Contents
Configuring Advanced Features...........................................131
xiii
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Configuring Audio Features ..................................................199
Configuring Security Features ...............................................211
Upgrading Firmware .............................................................225
Resource Files ........................................................................229
Troubleshooting .....................................................................241
xiv
Table of Contents
What is the difference between a remote phone book and a local phone book? ....... 249
What is the difference among user name, register name and display name? ............. 249
Why does the IP phone use DOB format logo file instead of popular BMP, JPG and so on?
What are the main differences among SIP-T28P, IP-T26P, SIP-T22P, SIP-T21P, SIP-T20P and
Appendix ...............................................................................255
xv
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Index ......................................................................................477
xvi
Product Overview
This chapter contains the following information about IP phones:
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.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
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.
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.
2
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.
User Agent Client (UAC)
The UAC is an application that initiates up to six feasible SIP requests to the UAS. The six requests issued by the UAC are: INVITE, ACK, OPTIONS, BYE, CANCEL and REGISTER.
When the SIP session is being initiated by the UAC SIP component, the UAC determines the information essential for the request, which is the protocol, the port and the IP address of the UAS to which the request is being sent. This information can be dynamic and will make it challenging to put through a firewall. For this reason, it may be recommended to open the specific application type on the firewall. The UAC is also capable of using the information in the request URI to establish the course of the SIP request to its destination, as the request URI always specifies the host which is essential.
The port and protocol are not always specified by the request URI. Thus if the request does not specify a port or protocol, a default port or protocol is contacted. It may be
Product Overview preferential to use this method when not using an application layer firewall. Application layer firewalls like to know what applications are flowing though which ports and it is possible to use content types of other applications other than the one you are trying to let through what has been denied.
User agent server (UAS)
UAS is a server that hosts the application responsible for receiving the SIP requests from a UAC, and on reception it returns a response to the request back to the UAC. The UAS may issue multiple responses to the UAC, not necessarily a single response.
Communication between UAC and UAS is client/server and peer-to–peer.
Typically, a SIP endpoint is capable of functioning as both a UAC and a UAS, but it functions only as one or the other per transaction. Whether the endpoint functions as a
UAC or a UAS depends on the UA that initiates the request.
This section introduces SIP IP phone models. IP phones are endpoints in the overall network topology, which are designed to interoperate with other compatible equipments including application servers, media servers, internet-working gateways, voice bridges, and other endpoints. 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.
IP phones provide a powerful and flexible IP communication solution for Ethernet TCP/IP networks, delivering excellent voice quality. The high-resolution graphic display supplies content in multiple languages for system status, call log and directory access.
IP phones also support advanced functionalities, including LDAP, Busy Lamp Field, Sever
Redundancy and Network Conference.
The following IP phone models are described:
SIP-T28P
SIP-T26P
SIP-T22P
SIP-T21P
SIP-T20P
SIP-T19P
IP phones comply with the SIP standard (RFC 3261), and they can only be used within a network that supports this model of phone.
In order to operate as SIP endpoints in your network successfully, IP phones must meet the following requirements:
A working IP network is established.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Routers are configured for VoIP.
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.
This section lists the available physical features of IP phones.
SIP-T28P
4
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 DSS keys
- 1xRJ9 (4P4C) handset port
- 1xRJ9 (4P4C) headset port
- 2xRJ45 10/100Mbps 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)
Product Overview
SIP-T26P
Physical Features:
- TI TITAN chipset and TI voice engine
- 132x64 graphic LCD
- 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated
- HD Voice: HD Codec, HD Handset, HD Speaker
- 45 keys including 13 DSS keys
- 1xRJ9 (4P4C) handset port
- 1xRJ9 (4P4C) headset port
- 2xRJ45 10/100Mbps Ethernet ports
- 1XRJ12 (6P6C) expansion module port
- 16 LEDs: 1xpower, 3xline, 1xmessage, 1xheadset, 10xmemory
- Power adapter: AC 100~240V input and DC 5V/1.2A output
- Power over Ethernet (IEEE 802.3af)
5
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
SIP-T22P
6
Physical Features:
- TI TITAN chipset and TI voice engine
- 132x64 graphic LCD
- 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated
- HD Voice: HD Codec, HD Handset, HD Speaker
- 32 keys including 4 soft keys
- 1xRJ9 (4P4C) handset port
- 1xRJ9 (4P4C) headset port
- 2xRJ45 10/100Mbps Ethernet ports
- 5 LEDs: 1xpower, 3xline, 1xmessage
- Power adapter: AC 100~240V input and DC 5V/1.2A output
- Power over Ethernet (IEEE 802.3af)
- Wall Mount
Product Overview
SIP-T21P
Physical Features:
- 132x64 graphic LCD
- 2 VoIP accounts
- 31 keys including 4 soft keys
- 4 LEDs: 1xpower, 2xline, 1xmessage
- HD Voice: HD Codec, HD Handset, HD Speaker
- 1xRJ9 (4P4C) handset port
- 1xRJ9 (4P4C) headset port
- 2xRJ45 10/100Mbps Ethernet ports
- Power adapter: AC 100~240V input and DC 5V/600mA output
- Power over Ethernet (IEEE 802.3af)
- Wall Mount
7
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
SIP-T20P
Physical Features:
- TI TITAN chipset and TI voice engine
- 3-line LCD consists of an icon line and two 15-character lines
- 2 VoIP accounts, BroadSoft/Avaya/Asterisk validated
- HD Voice: HD Codec, HD Handset, HD Speaker
- 31 keys including 9 function keys
- 1xRJ9 (4P4C) handset port
- 1xRJ9 (4P4C) headset port
- 2xRJ45 10/100Mbps Ethernet ports
- 4 LEDs: 1xpower, 2xline, 1xmessage
- Power adapter: AC 100~240V input and DC 5V/1.2A output
- Power over Ethernet (IEEE 802.3af)
- Wall Mount
8
SIP-T19P
Physical Features:
- 132x64 graphic LCD
- Single VoIP account
- 29 keys including 4 soft keys
- 1xRJ9 (4P4C) handset port
- 1xRJ9 (4P4C) headset port
- 2xRJ45 10/100Mbps Ethernet ports
- 1 LED: 1xpower
- Power adapter: AC 100~240V input and DC 5V/600mA output
- Power over Ethernet (IEEE 802.3af)
- Wall Mount
Product Overview
9
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
10
In addition to physical features introduced above, IP phones also support the following key features when running the latest firmware:
Phone Features
- Call Options: emergency call, call waiting, call hold, call mute, call forward, call transfer, call pickup, conference.
- Basic Features: DND, phone lock, auto redial, live dialpad, dial plan, hotline, caller identity, auto answer.
- Advanced Features: BLF, server redundancy, distinctive ring tones, remote phone book (not applicable to SIP-T20P IP phones), LDAP (not applicable to
SIP-T19P and SIP-T20P IP phones), 802.1X authentication.
Codecs and Voice Features
- Wideband codec: G.722
- Narrowband codec: G.711, G.723, G.726, G.729AB, iLBC
- 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
- VLAN assignment: LLDP/Static/DHCP
- Bridge/Router mode for PC port (Router mode is not applicable to SIP-T19P and
SIP-T21P IP phones)
- TFTP/DHCP/PPPoE client
- HTTP/HTTPS server
- DNS client
- NAT/DHCP server
- IPv6 support
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
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
Product Overview
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
12
Getting Started
This chapter provides basic information and installation instructions of IP phones.
This chapter provides the following sections:
Initialization Process Overview
Configuring Basic Network Parameters
This section introduces how to install IP phones with components in packaging contents.
1. Attach the stand
2. Connect the handset and optional headset
3. Connect the network and power
Note A headset is not included in packaging contents.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
1) Attach the stand:
SIP-T28P/T26P
SIP-T22P/T21P/T20P
14
SIP-T19P
Getting Started
2) Connect the handset and optional headset:
SIP-T28P/T26P
SIP-T22P/T21P/T20P/T19P
3) Connect the network and power:
AC power
Power over Ethernet (PoE)
AC Power
To connect the AC power and network:
1. Connect the DC plug of the power adapter to the DC5V port on the IP phone and connect the other end of the power adapter into an electrical power outlet.
2. Connect the included or a standard Ethernet cable between the Internet port on the IP phone and the one on the wall or switch/hub device port.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Power over Ethernet
With the included or a regular Ethernet cable, IP phones can be powered from a
PoE-compliant switch or hub.
To connect the PoE:
1. Connect the Ethernet cable between the Internet port on the IP phone and an available port on the in-line power switch/hub.
Note If in-line power switch/hub is provided, you don’t need to connect the phone to the power adapter. Make sure the switch/hub is PoE-compliant.
The IP phone can also share the network with another network device such as a PC
(personal computer). It is an optional connection.
Important! Do not unplug or remove the power while the IP phone is updating firmware and configurations.
16
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). The IP phone can then proceed with the DHCP request for its network settings (if using DHCP).
Getting Started
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
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 24 .
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.
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.
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
After connected to the power and network, the IP phone begins the initializing process by cycling through the following steps:
1. The power indicator LED illuminates.
17
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
2. The message “Initializing, Please Wait” appears on the LCD screen when the IP phone starts up.
3. The main LCD screen displays the following:
Time and date
Soft key labels (not applicable to SIP-T20P IP phones)
4. Press the OK key to check the IP phone status, the LCD screen displays the valid IP address, MAC address, firmware version, etc.
If the IP phone has successfully passed through these steps, it starts up properly and is ready for use.
You can use the following methods to set up and configure IP phones:
The following sections describe how to configure IP phones using each method above.
An administrator or a user can configure and use IP phones via phone user interface.
Access to specific features is restricted to the administrator. The default password is
“admin“(case-sensitive). Not all features are available on phone user interface.
An administrator or a user can configure IP phones via web user interface. The default user name and password for the administrator to log into the web user interface are both “admin” (case-sensitive). Almost all features are available on web user interface.
IP phones support both HTTP and HTTPS protocols for accessing the web user interface.
For more information, refer to Web Server Type on page 117 .
18
You can deploy IP phones using configuration files. There are two configuration files both of which are CFG formatted. We call them Common CFG file and MAC-Oriented
CFG file. A Common CFG file will be effectual for all IP phones of the same model.
However, a MAC-Oriented CFG file will only be effectual for a specific IP phone. The
Common CFG file has a fixed name for each IP phone model, while the MAC-Oriented
Getting Started
CFG file is named after the MAC address of the IP phone. For example, if the MAC address of a SIP-T22P IP phone is 001565113af8, names of these two configuration files must be: y000000000005.cfg and 001565113af8.cfg.
The name of the Common CFG file for each IP phone model is:
SIP-T28P: y000000000000.cfg
SIP-T26P: y000000000004.cfg
SIP-T22P: y000000000005.cfg
SIP-T21P: y000000000034.cfg
SIP-T20P: y000000000007.cfg
SIP-T19P: y000000000031.cfg
In order to deploy IP phones using the configuration files (<y0000000000xx>.cfg and
<MAC>.cfg), you need to use a text-based editing application to edit configuration files, and store configuration files to a provisioning server. IP phones support downloading configuration files using any of the following protocols: FTP, TFTP, HTTP and
HTTPS.
IP phones can obtain the address of the provisioning server during startup through one of the following processes: Zero Touch, PnP, DHCP Options and Phone Flash. Then IP phones download configuration files from the provisioning server, resolve and update the configurations written in configuration files. This entire process is called auto provisioning. For more information on auto provisioning, refer to
Yealink_SIP-T2
Series_T19P_T4_Series_IP_Phones_Auto_Provisioning_Guide
.
When modifying parameters, learn the following:
Parameters in configuration files override those stored in the IP phone’s flash memory.
The .cfg extension of configuration files must be in lowercase.
Each line in a configuration file must use the following format and adhere to the following rules: variable-name = value
- Associate only one value with one variable.
- Separate variable name and value with equal sign.
- Set only one variable per line.
- Put the variable and value on the same line, and do not break the line.
- Comment the variable on a separated line. Use the pound (#) delimiter to distinguish the comments.
IP phones can accept two sources of configuration data:
Downloaded from configuration files
Changed on the phone user interface or the web user interface
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
The latest values configured on the IP phone take effect finally.
20
Icons associated with different features may appear on the LCD screen. The following table provides a description for each icon on IP phones.
SIP-T28P SIP-T26P SIP-T22P SIP-T21P SIP-T20P SIP-T19P
/
/
Description
Network is unavailable
Registered successfully
Registration failed
/ Registering
Hands-free speakerphone mode
Handset mode
Headset mode
Voice Mail
/
/
Text Message
Auto Answer
Do Not Disturb
Call
Forward/Forwar ded Calls
Call Hold
Call Mute
Getting Started
SIP-T28P SIP-T26P SIP-T22P SIP-T21P SIP-T20P SIP-T19P
/
Description
Ringer volume is
0
Phone Lock
/
/
/
/
/
Received Calls
Placed Calls
/
/
/
/
/
Missed Calls
Recording box is full
A call cannot be recorded
Recording starts successfully
Recording cannot be started
Recording cannot be stopped
This section describes how to configure basic network parameters for the IP phone.
Note This section mainly introduces IPv4 network parameters. IP phones also support IPv6. For more information on IPv6, refer to
DHCP (Dynamic Host Configuration Protocol) is a network protocol used to dynamically allocate network parameters to network hosts. The automatic allocation of network parameters to hosts eases the administrative burden of maintaining an IP network. IP phones comply with the DHCP specifications documented in RFC 2131. If using DHCP, IP phones connected to the network become operational without having to be manually assigned IP addresses and additional network parameters. Static DNS address(es) can
21
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones be configured and used when DHCP is enabled.
DHCP Option
DHCP provides a framework for passing information to TCP/IP network devices. Network and other control information are carried in tagged data items that are stored in the options field of the DHCP message. The data items themselves are also called options.
DHCP can be initiated by simply connecting the IP phone with the network. IP phones broadcast DISCOVER messages to request the network information carried in DHCP options, and the DHCP server responds with specific values in corresponding options.
The following table lists common DHCP options supported by IP phones.
Parameter
Subnet Mask
Time Offset
Router
Time Server
Domain Name
Server
Log Server
Host Name
Domain Server
DHCP Option
1
2
3
4
6
7
12
15
28
Description
Specify the client’s subnet mask.
Specify the offset of the client's subnet in seconds from Coordinated Universal Time
(UTC).
Specify a list of IP addresses for routers on the client’s subnet.
Specify a list of time servers available to the client.
Specify a list of domain name servers available to the client.
Specify a list of MIT-LCS UDP servers available to the client.
Specify the name of the client.
Specify the domain name that client should use when resolving hostnames via DNS.
Specify the broadcast address in use on the client's subnet.
Broadcast
Address
Network Time
Protocol
Servers
Vendor-Specific
Information
Vendor Class
Identifier
42
43
60
Specify a list of NTP servers available to the client by IP address.
Identify the vendor-specific information.
Identify the vendor type.
TFTP Server 66
Identify a TFTP server when the 'sname' field in the DHCP header has been used for DHCP
22
Getting Started
Parameter
Name
Boot file Name
DHCP Option
67
Description options.
Identify a boot file when the 'file' field in the
DHCP header has been used for DHCP options.
Procedure
DHCP can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure DHCP on the IP phone.
Configure static DNS address when DHCP is used.
For more information, refer to
Configure DHCP on the IP phone.
Configure static DNS address when DHCP is used.
Navigate to: http://<phoneIPAddress>/servlet
?p=network&q=load
Phone User Interface Configure DHCP on the IP phone.
To configure DHCP via web user interface:
1. Click on Network->Basic.
2. In the IPv4 Config block, mark the DHCP radio box.
23
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
3. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
4. Click OK to reboot the IP phone.
To configure static DNS address when DHCP is used via web user interface:
1. Click on Network->Basic.
2. In the IPv4 Config block, mark the DHCP radio box.
3. Mark the Static DNS radio box.
4. Enter the desired values in the Primary DNS and Secondary DNS fields.
5. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
6. Click OK to reboot the IP phone.
To configure DHCP via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN
Port->IPv4.
2. Press or to highlight the DHCP IP Client field.
3. Press the Save soft key to accept the change.
The IP phone reboots automatically to make settings effective after a period of time.
24
If DHCP is disabled or IP phones cannot obtain network parameters from the DHCP server, you need to configure them manually. The following parameters should be configured for IP phones to establish network connectivity:
IP Address
Subnet Mask
Getting Started
Default Gateway
Primary DNS
Secondary DNS
Procedure
Network parameters can be configured manually using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure network parameters of the IP phone manually.
For more information, refer to
Static Network Settings on page
Configure network parameters of the IP phone manually.
Navigate to: http://<phoneIPAddress>/servlet
?p=network&q=load
Configure network parameters of the IP phone manually.
To configure the IP address mode via web user interface:
1. Click on Network->Basic.
2. Select desired value from the pull-down list of Mode (IPv4/IPv6).
3. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
4. Click OK to reboot the IP phone.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
To configure a static IPv4 address via web user interface:
1. Click on Network->Basic.
2. In the IPv4 Config block, mark the Static IP Address radio box.
3. Enter the desired values in the IP Address, Subnet Mask, Gateway, Primary DNS and Secondary DNS fields.
4. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
5. Click OK to reboot the IP phone.
To configure the IP address mode via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN
Port.
2. Press or to select IPv4, IPv6 or IPv4&IPv6 from the IP Mode field.
3. Press the Save soft key to accept the change.
The IP phone reboots automatically to make settings effective after a period of time.
To configure a static IPv4 address via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN
Port->IPv4->Static IP Client.
2. Enter the desired values in the IPv4, Subnet Mask, Default Gateway, Pri DNS and
Sec DNS fields.
3. Press the Save soft key to accept the change.
The IP phone reboots automatically to make settings effective after a period of time.
Note Using the wrong network parameters may result in inaccessibility of your phone and may also have an impact on your network performance. For more information on these parameters, contact your network administrator.
26
Getting Started
PPPoE (Point-to-Point Protocol over Ethernet) is a network protocol used by Internet
Service Providers (ISPs) to provide Digital Subscriber Line (DSL) high speed Internet services. PPPoE allows an office or building-full of users to share a common DSL connection to the Internet. PPPoE connection is supported by the IP phone Internet port.
Contact your ISP for the PPPoE user name and password.
Procedure
PPPoE can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure PPPoE on the IP phone.
For more information, refer to
Configure PPPoE on the IP phone.
Navigate to: http://<phoneIPAddress>/servlet
?p=network&q=load
Phone User Interface Configure PPPoE on the IP phone.
To configure PPPoE via web user interface:
1. Click on Network->Basic.
2. In the IPv4 Config block, mark the PPPoE radio box.
3. Enter the user name and password in corresponding fields.
4. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
27
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
5. Click OK to reboot the IP phone.
To configure PPPoE via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN
Port->IPv4->PPPoE IP Client.
2. Enter the user name and password in corresponding fields.
3. Press the Save soft key to accept the change.
The IP phone reboots automatically to make settings effective after a period of time.
Two Ethernet ports on the back of the IP phone: Internet port and PC port. Three optional methods of transmission configuration for IP phone Internet or PC Ethernet ports:
Auto-negotiation
Half-duplex
Full-duplex
Auto-negotiation is configured for both Internet and PC ports on the IP phone by default.
Auto-negotiation
Auto-negotiation means that two connected devices choose common transmission parameters (e.g., speed and duplex mode) to transmit voice or data over Ethernet. This process entails devices first sharing transmission capabilities and then selecting the highest performance transmission mode supported by both. You can configure the
Internet port and PC port on the IP phone to automatically negotiate during the transmission.
28
Getting Started
Half-duplex
Half-duplex transmission refers to transmitting voice or data in both directions, but in one direction at a time; this means one device can send data on the line, but not receive data simultaneously. You can configure the half-duplex transmission on both
Internet port and PC port for the IP phone to transmit in 10Mbps or 100Mbps.
Full-duplex
Full-duplex transmission refers to transmitting voice or data in both directions at the same time; this means one device can send data on the line while receiving data. You can configure the full-duplex transmission on both Internet port and PC port for the IP phone to transmit in 10Mbps or 100Mbps.
Procedure
The transmission methods of Ethernet ports can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Configure the transmission methods of Ethernet ports.
For more information, refer to
29
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Local Web User Interface
Configure the transmission methods of Ethernet ports.
Navigate to: http://<phoneIPAddress>/servlet
?p=network-adv&q=load
To configure the transmission methods of Ethernet ports via web user interface:
1. Click on Network->Advanced.
2. Select the desired value from the pull-down list of WAN Port Link.
3. Select the desired value from the pull-down list of PC Port Link.
4. Click Confirm to accept the change.
The PC port on the back of the IP phone is used to connect a PC, which can be configured in one of two modes:
Bridge: The IP phone functions as a bridge, and the connected PC appears on the network as a stand-alone device with its own IP address.
Router: The IP phone functions as a router, and provides a DHCP service for the connected PC.
Note
The router mode is not applicable to SIP-T19P and SIP-T21P IP phones.
30
Getting Started
Procedure
PC port mode can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the PC port mode.
For more information, refer to PC
Configure the PC port mode.
Navigate to: http://<phoneIPAddress>/servlet
?p=network-pcport&q=load
Phone User Interface Configure the PC port mode.
To configure the PC port mode via web user interface:
1. Click on Network->PC Port.
2. Select the desired value from the pull-down list of PC Port Active.
3. Mark the desired radio box.
If you mark the As Router radio box, you can configure the IP address for the PC port and configure DHCP for the PC attached to the PC port.
1) Enter the IP address in the IP Address field.
2) Enter subnet mask in the Subnet Mask field.
3) Select the desired value from the pull-down list of Enable DHCP Server.
4) Enter the start IP address in the Start IP Address field.
5) Enter the end IP address in the End IP Address field.
4. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
5. Click OK to reboot the IP phone.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
To configure the PC port mode via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->PC
Port.
2. Select the desired mode.
If you select Router, you can configure the IP address for the PC port and configure
DHCP for the PC attached to the PC port.
1) Enter the IP address in the IPv4 field.
2) Enter the subnet mask in the Subnet Mask field.
3) Press or to highlight the DHCP Server field, and then press the Enter soft key.
4) Select the desired value from the Server Status field.
5) Enter the start IP address in the Start IP field.
6) Enter the end IP address in the End IP field.
3. Press the Save soft key to accept the change.
The IP phone reboots automatically to make settings effective after a period of time.
32
Regular expression, often called a pattern, is an expression that specifies a set of strings.
A regular expression provides a concise and flexible means to “match” (specify and recognize) strings of text, such as particular characters, words, or patterns of characters.
Regular expression is used by many text editors, utilities, and programming languages to search and manipulate text based on patterns.
Regular expression can be used to define IP phone dial plan. Dial plan is a string of characters that governs the way for IP phones to process the inputs received from the IP phone’s keypads. IP phones support the following dial plan features:
You need to know the following basic regular expression syntax when creating dial plan:
.
x
The dot “.” can be used as a placeholder or multiple placeholders for any string. Example:
“12.” would match “123”, “1234”, “12345”, “12abc”, etc.
The “x” can be used as a placeholder for any character. Example:
“12x” would match “121”, “122”, “123”, “12a”, etc.
Getting Started
-
,
[]
()
$
The dash “-” can be used to match a range of characters within the brackets. Example:
“[5-7]” would match the number “5”, ”6” or ”7”.
The comma “,” can be used as a separator within the bracket.
Example:
“[2,5,8]” would match the number ”2”, “5” or “8”.
The square bracket "[]" can be used as a placeholder for a single character which matches any of a set of characters. Example:
"91[5-7]1234"would match “9151234”, “9161234”, “9171234”.
The parenthesis "( )" can be used to group together patterns, for instance, to logically combine two or more patterns. Example:
"([1-9])([2-7])3" would match “923”, “153”, “673”, etc.
The “$” followed by the sequence number of a parenthesis means the characters placed in the parenthesis. The sequence number stands for the corresponding parenthesis. Example:
A replace rule configuration, Prefix: "001(xxx)45(xx)", Replace:
"9001$145$2". When you dial out "0012354599" on your phone, the IP phone will replace the number with "90012354599". “$1” means 3 digits in the first parenthesis, that is, “235”. “$2” means 2 digits in the second parenthesis, that is, “99”.
Replace rule is an alternative string that replaces the numbers entered by the user. IP phones support up to 100 replace rules, which can be created either one by one or in batch using a replace rule template. For more information on the replace rule template,
refer to Replace Rule Template on page 229 .
Procedure
Replace rule can be created using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Create the replace rule for the IP phone.
For more information, refer to Dial
Create the replace rule for the IP phone.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-dialplan&q=load
33
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
To create a replace rule via web user interface:
1. Click on Settings->Dial Plan->Replace Rule.
2. Enter the string in the Prefix field.
3. Enter the string in the Replace field.
4. Enter the desired line ID in the Account field or leave it blank.
If you leave this field blank or enter 0, the replace rule will apply to all accounts on the IP phone.
5. Click Add to add the replace rule.
34
Dial-now is a string used to match numbers entered by the user. When entered numbers match the predefined dial-now rule, the IP phone will automatically dial out the numbers without pressing the send key. IP phones support up to 100 dial-now rules, which can be created either one by one or in batch using a dial-now rule template. For
more information on the dial-now template, refer to Dial-now Template on page 230 .
Delay Time for Dial-now Rule
The IP phone will automatically dial out the entered number, which matches the dial-now rule, after a specified period of time.
Procedure
Dial-now rule can be created using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Create the dial-now rule for the IP phone.
Configure the delay time for the
Getting Started
Local Web User Interface dial-now rule.
For more information, refer to Dial
Create the dial-now rule for the IP phone.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-dialnow&q=load
Configure the delay time for the dial-now rule.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
To create a dial-now rule via web user interface:
1. Click on Settings->Dial Plan->Dial-now.
2. Enter the desired value in the Rule field.
3. Enter the desired line ID in the Account field or leave it blank.
If you leave this field blank or enter 0, the dial-now rule will apply to all accounts on the IP phone.
4. Click Add to add the dial-now rule.
To configure the delay time for the dial-now rule via web user interface:
1. Click on Features->General Information.
35
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
2. Enter the desired time within 1-14 (in seconds) in the Time-Out for Dial-Now Rule field.
3. Click Confirm to accept the change.
36
Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country. When entered numbers match the predefined area code rule, the IP phone will automatically add the area code before the numbers when dialing out them. IP phones only support one area code rule.
Procedure
Area code rule can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Create the area code rule and specify the maximum and minimum lengths of entered numbers.
For more information, refer to Dial
Create the area code rule and specify the maximum and minimum lengths of entered numbers.
Navigate to: http://<phoneIPAddress>/servlet
Getting Started
?p=settings-areacode&q=load
To configure an area code rule via web user interface:
1. Click on Settings->Dial Plan->Area Code.
2. Enter the desired values in the Code, Min Length (1-15) and Max Length (1-15) fields.
3. Enter the desired line ID in the Account field or leave it blank.
If you leave this field blank or enter 0, the area code rule will apply to all accounts on the IP phone.
4. Click Confirm to accept the change.
Block out rule prevents users from dialing out specific numbers. When entered numbers match the predefined block out rule, the LCD screen prompts “Forbidden Number”. IP phones support up to 10 block out rules.
Procedure
Block out rule can be created using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Create the block out rule for the
IP phone.
For more information, refer to Dial
Create the block out rule for the desired line.
Navigate to: http://<phoneIPAddress>/servlet
37
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
?p=settings-blackout&q=load
To create a block out rule via web user interface:
1. Click on Settings->Dial Plan->Block Out.
2. Enter the desired value in the BlockOut Number field.
3. Enter the desired line ID in the Account field or leave it blank.
If you leave this field blank or enter 0, the block out rule will apply to all accounts on the IP phone.
4. Click Confirm to add the block out rule.
38
Configuring Basic Features
This chapter provides information for making configuration changes for the following basic features:
39
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Transfer on Conference Hang Up
Calling Line Identification Presentation
Connected Line Identification Presentation
40
Power indicator LED indicates power status and phone status. There are six configuration options for power indicator LED:
Common Power Light On
Common Power Light On allows the power indicator LED to be turned on.
Ring Power Light Flash
Ring Power Light Flash allows the power indicator LED to flash when the IP phone receives an incoming call. If this option is disabled, the status of the power indicator LED is determined by the option “Common Power Light On”.
Voice/Text Mail Power Light Flash
Voice/Text Mail Power Light Flash allows the power indicator LED to flash when the IP phone receives a voice mail or a text message. If this option is disabled, the status of the power indicator LED is determined by the option “Common Power Light On”.
Mute Power Light Flash
Mute Power Light Flash allows the power indicator LED to flash when a call is mute. If
Configuring Basic Features this option is disabled, the status of the power indicator LED is determined by the option
“Common Power Light On”.
Hold/Held Power Light Flash
Hold/Held Power Light Flash allows the power indicator LED to flash when a call is placed on hold or is held. If this option is disabled, the status of the power indicator LED is determined by the option “Common Power Light On”.
Talk/Dial Power Light On
Talk/Dial Power Light On allows the power indicator LED to be turned on when the IP phone is busy. If this option is disabled, the status of the power indicator LED is determined by the option “Common Power Light On”.
Procedure
Power indicator LED can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the power indicator
LED.
For more information, refer to
on page
Configure the power indicator
LED.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-powerled&q=load
To configure the power Indicator LED via web user interface:
1. Click on Features->Power LED.
2. Select the desired value from the pull-down list of Common Power Light On.
3. Select the desired value from the pull-down list of Ring Power Light Flash
4. Select the desired value from the pull-down list of Voice/Text Mail Power Light Flash.
5. Select the desired value from the pull-down list of Mute Power Light Flash.
6. Select the desired value from the pull-down list of Hold/Held Power Light Flash.
41
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
7. Select the desired value from the pull-down list of Talk/Dial Power Light On.
8. Click Confirm to accept the change.
42
Contrast determines the readability of the texts displayed on the LCD screen. Adjusting the contrast to a comfortable level can optimize the screen viewing experience. When configured properly, contrast allows users to read the LCD’s display with minimal eyestrain. The contrast of the LCD screen is only applicable to SIP-T19P, SIP-T21P and
SIP-T28P IP phones, and EXP39 connected to SIP-T26P and SIP-T28P IP phones.
Procedure
Contrast can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure the contrast of the LCD screen.
For more information, refer to
on page
Configure the contrast of the LCD screen.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-preference&q=load
Configure the contrast of the LCD screen.
To configure contrast via web user interface:
1. Click on Settings->Preference.
Configuring Basic Features
2. Select the desired value from the pull-down list of Contrast.
3. Click Confirm to accept the change.
To configure contrast via phone user interface (applicable to SIP-T28P IP phones and
EXP39 connected to SIP-T26P and SIP-T28P IP phones):
1. Press Menu->Settings->Basic Settings->Display->Contrast.
2. Press or , or the Switch soft key to increase or decrease the intensity of contrast.
The default contrast level is 6.
3. Press the Save soft key to accept the change.
Note Before you adjust the LCD’s contrast of the expansion module, make sure the expansion module has been connected to the IP phone.
To configure contrast via phone user interface (applicable to SIP-T19P and SIP-T21P IP phones):
1. Press Menu->Settings->Basic Setting->Contrast.
2. Press or , or the Switch soft key to increase or decrease the intensity of contrast.
The default contrast level is 6.
3. Press the Save soft key to accept the change.
Backlight determines the brightness of the LCD screen display, allowing users to read easily in dark environments. Backlight time specifies the delay time to turn off the backlight when the IP phone is inactive. Backlight time is applicable to SIP-T22P, SIP-T26P and SIP-T28P IP phones, and EXP39 connected to SIP-T26P and SIP-T28P IP phones.
Backlight turns off quickly if a short backlight time is configured, this may not give users
43
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones enough time to read messages. Backlight active level is used to adjust the backlight intensity of the LCD screen. Backlight active level is only applicable to SIP-T28P IP phones and the connected EXP39.
You can configure the backlight time as one of the following types:
Always Off: Backlight is turned off permanently.
Always On: Backlight is turned on permanently.
15, 30, 60, 120, 300, 600 or 1800: Backlight is turned off when the IP phone is inactive after a preset period of time (in seconds), but it is automatically turned on if the status of the IP phone changes or any key is pressed.
The following table lists available methods and configuration options to configure the backlight of each phone model.
Phone Model
SIP-T28P
SIP-T26P
SIP-T22P
Configuration Methods
Configuration Files
Web User Interface
Phone User Interface
Configuration Files
Web User Interface
Phone User Interface (only applicable to the connected
EXP39)
Configuration Files
Web User Interface
Configuration Options
Backlight Active Level
Backlight Time
Backlight Active Level (only applicable to the connected
EXP39)
Backlight Time
Backlight Time
Procedure
Backlight can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure the backlight of the
LCD screen.
For more information, refer to
Configure the backlight of the
LCD screen.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-preference&q=load
Configure the backlight of the
LCD screen.
44
Configuring Basic Features
To configure backlight via web user interface:
1. Click on Settings->Preference.
2. Select the desired value from the pull-down list of Backlight Active Level (only applicable to SIP-T28P IP phones and the connected EXP39).
3. Select the desired value from the pull-down list of Backlight Time (seconds).
4. Click Confirm to accept the change.
To configure backlight via phone user interface (only applicable to SIP-T28P IP phones and EXP39 connected to SIP-T26P and SIP-T28P IP phones):
1. Press Menu->Settings->Basic Settings->Display->Backlight.
2. Press or , or the Switch soft key to select the desired level from the Active
Level field.
3. Press or , or the Switch soft key to select the desired type from the
Backlight Time field.
4. Press the Save soft key to accept the change.
Note Before you adjust the LCD’s backlight of expansion module, make sure the expansion module has been connected to the IP phone.
Some menu options are protected by two privilege levels, user and administrator, each with its own password. When logging into the web user interface, you need to enter the user name and password to access various menu options.
A user or an administrator can change the user password. The default user password is
“user”. For security reasons, the user or administrator should change the default user password as soon as possible.
45
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Procedure
User password can be changed using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Change the user password of the
IP phone.
For more information, refer to
Change the user password of the
IP phone.
Navigate to: http://<phoneIPAddress>/servlet
?p=security&q=load
To change the user password via web user interface:
1. Click on Security->Password.
2. Select user from the pull-down list of User Type.
3. Enter new password in the New Password and Confirm Password fields.
The new password should be complex and contains at least 6 characters, where at least one character is numeric, and one character is alphabetic. Valid characters contain A-Z, a-z, 0-9,#,!,@,-,.,*,+ and $.
4. Click Confirm to accept the change.
Note If logging into the web user interface of the phone with the user credential, you need to enter the old user password in the Old Password field.
46
Advanced menu options are strictly used by administrators. Users can configure them only if they have administrator privileges. The administrator password can only be changed by an administrator. The default administrator password is “admin”. For security reasons, the administrator should change the default administrator password as soon as possible.
Configuring Basic Features
Procedure
Administrator password can be changed using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Change the administrator password.
For more information, refer to
on page
Change the administrator password.
Navigate to: http://<phoneIPAddress>/servlet
?p=security&q=load
Change the administrator password.
To change the administrator password via web user interface:
1. Click on Security->Password.
2. Select admin from the pull-down list of User Type.
3. Enter the current administrator password in the Old Password field.
4. Enter new password in the New Password and Confirm Password fields.
The new password should be complex and contains at least 6 characters, where at least one character is numeric, and one character is alphabetic. Valid characters contain A-Z, a-z, 0-9,#,!,@,-,.,*,+ and $.
5. Click Confirm to accept the change.
To change the administrator password via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Set Password.
2. Enter the current administrator password in the Current PWD field.
3. Enter new password in the New PWD field and Confirm PWD field.
4. Press the Save soft key to accept the change.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
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Phone lock is used to lock the IP phone to prevent it from unauthorized use. Once the
IP phone is locked, a user must enter the password to unlock it. IP phones offer three types of phone lock: Menu Key, Function Keys and All Keys. The IP phone will not be locked immediately after the phone lock type is configured. One of the following steps is also needed:
- Long press the pound key when the IP phone is idle.
- Press the keypad lock key (if configured) when the IP phone is idle.
In addition to the above steps, you can configure the IP phone to automatically lock the keypad after a period of time.
Procedure
Phone lock can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the type of phone lock.
Change the unlock PIN.
Configure the IP phone to automatically lock the keypad after a time interval.
For more information, refer to
on page
Assign a keypad lock key.
For more information, refer to
Configure the type of phone lock.
Change the unlock PIN.
Configure the IP phone to automatically lock the keypad after a time interval.
Navigate to: http://<phoneIPAddress>/servl et?p=features-phonelock&q=lo ad
Assign a keypad lock key.
Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&q=load&model=
Configuring Basic Features
Phone User Interface
0
Configure the type of phone lock.
Assign a keypad lock key.
To configure phone lock via web user interface:
1. Click on Features->Phone Lock.
2. Select the desired type from the pull-down list of Keypad Lock Type.
3. Enter the unlock PIN in the Phone Unlock PIN (0~15 Digit) field.
4. Enter the desired time in the Phone Lock Time Out (0~3600s) field.
5. Click Confirm to accept the change.
To configure a keypad lock key via web user interface:
1. Click on DSSKey->Memory Key (Line Key or Programable Key).
SIP-T19P IP phones only support programable keys and SIP-T22P/T21P/T20P IP phones only support line keys and programable keys.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
2. In the desired DSS key field, select Keypad Lock from the pull-down list of Type.
3. Click Confirm to accept the change.
To configure the type of phone lock via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Phone
Settings->Keypad Lock.
2. Press or , or the Switch soft key to select the desired type from the Keypad
Lock field.
3. Press the Save soft key to accept the change.
To change the unlock PIN via phone user interface:
1. Press Menu->Settings->Basic Settings->Phone Unlock PIN.
2. Enter the current unlock PIN in the Current PIN field.
3. Enter the new unlock PIN in the New PIN field.
4. Enter the new unlock PIN again in the Confirm PIN field.
5. Press the Save soft key to accept the change.
To configure a keypad lock key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Keypad Lock from the Type field.
4. Press the Save soft key to accept the change.
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IP phones maintain a local clock and calendar. Time and date are displayed on the idle screen of IP phones. Time and date are synced automatically from the NTP server by default. The NTP server can be obtained by DHCP or configured manually. If IP phones cannot obtain the time and date from the NTP server, you need to manually configure
Configuring Basic Features them. The time and date display can use one of several different formats.
Time Zone
A time zone is a region on Earth that has a uniform standard time. It is convenient for areas in close commercial or other communication to keep the same time. When configuring the IP phone to obtain the time and date from the NTP server, you must set the time zone.
Daylight Saving Time
Daylight Saving Time (DST) is the practice of temporary advancing clocks during the summertime so that evenings have more daylight and mornings have less. Typically, clocks are adjusted forward one hour at the start of spring and backward in autumn.
Many countries have used the DST at various times, details vary by location. The DST can be adjusted automatically from the time zone configuration. Typically, there is no need to change this setting.
The following table lists available configuration methods for time and date.
Time
Date
Option
Time Zone
Time Format
Date Format
Daylight Saving Time
Configuration Methods
Configuration Files
Web User Interface
Phone User Interface
Web User Interface
Phone User Interface
Configuration Files
Web User Interface
Phone User Interface
Web User Interface
Phone User Interface
Configuration Files
Web User Interface
Phone User Interface
Configuration Files
Web User Interface
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File <MAC>.cfg
Configure NTP by DHCP priority feature.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Local
Web User Interface
Phone User Interface
Configure the NTP server, time zone and DST.
Configure the time and date manually.
Configure the time and date formats.
For more information, refer to
Configure NTP by DHCP priority feature.
Configure the NTP server, time zone and DST.
Configure the time and date manually.
Configure the time and date formats.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-datetime&q=load
Configure the NTP server and time zone.
Configure the time and date manually.
Configure the time and date formats.
To configure NTP by DHCP priority feature via web user interface:
1. Click on Settings->Time & Date.
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Configuring Basic Features
2. Select the desired value from the pull-down list of NTP By DHCP Priority.
3. Click Confirm to accept the change.
To configure the NTP server, time zone and DST via web user interface:
1. Click on Settings->Time & Date.
2. Select Disabled from the pull-down list of Manual Time.
3. Select the desired time zone from the pull-down list of Time Zone.
4. Enter the domain names or IP addresses in the Primary Server and Secondary
Server fields respectively.
5. Enter the desired time interval in the Synchronism (15~86400s) field.
6. Select the desired value from the pull-down list of Daylight Saving Time.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
If you select Enabled, do one of the following:
- Mark the DST By Date radio box in the Fixed Type field.
Enter the start time in the Start Date field.
Enter the end time in the End Date field.
- Mark the DST By Week radio box in the Fixed Type field.
Select the desired values from the pull-down lists of DST Start Month, DST Start
Day of Week, DST Start Day of Week Last in Month, DST Stop Month, DST Stop
Day of Week and DST Stop Day of Week Last in Month.
Enter the desired time in the Start Hour of Day field.
Enter the desired time in the End Hour of Day field.
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7. Enter the desired offset time in the Offset (minutes) field.
Configuring Basic Features
8. Click Confirm to accept the change.
To configure the time and date manually via web user interface:
1. Click on Settings->Time & Date.
2. Select Enabled from the pull-down list of Manual Time.
3. Enter the time and date in the corresponding fields.
4. Click Confirm to accept the change.
To configure the time and date format via web user interface:
1. Click on Settings->Time & Date.
2. Select the desired value from the pull-down list of Time Format.
3. Select the desired value from the pull-down list of Date Format.
4. Click Confirm to accept the change.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
To configure the NTP server and time zone via phone user interface:
1. Press Menu->Settings->Basic Settings->Time & Date->SNTP Settings.
2. Press or , or the Switch soft key to select the time zone that applies to your area from the Time Zone field.
The default time zone is "+8 China(Beijing)".
3. Enter the domain names or IP addresses in the NTP Server1 and NTP Server2 fields respectively.
4. Press the Save soft key to accept the change.
To configure the time and date manually via phone user interface:
1. Press Menu->Settings->Basic Settings->Time & Date->Manual Settings.
2. Enter the date in the Date field.
3. Enter the time in the Time field.
4. Press the Save soft key to accept the change.
To configure the time and date formats via phone user interface:
1. Press Menu->Settings->Basic Settings->Time & Date->Time & Date Format.
2. Press or , or the Switch soft key to select the desired time format from the
Clock field.
3. Press or , or the Switch soft key to select the desired date format from the
Date Format field.
4. Press the Save soft key to accept the change.
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IP phones support multiple languages. Languages used on the phone user interface and web user interface can be specified respectively as required.
The following table lists languages supported by the phone user interface and the web user interface respectively.
Phone User Interface
English
German
Chinese_S (only applicable to
SIP-T19P and SIP-T21P IP phones)
Chinese_T (only applicable to
SIP-T19P and SIP-T21P IP phones)
French
Italian
Web User Interface
English
Chinese_S (only applicable to
SIP-T19P and SIP-T21P IP phones)
German
French (not applicable to SIP-T19P and SIP-T21P IP phones)
Italian
Portuguese (not applicable to
Configuring Basic Features
Phone User Interface
Portuguese
Polish
Spanish
Turkish
SIP-T19P and SIP-T21P IP phones)
Spanish (not applicable to
SIP-T19P and SIP-T21P IP phones)
Turkish
Web User Interface
Not all of supported languages are available for selection. Languages available for selection depend on language packs currently loaded to the IP phone. You can make languages available for use on the phone user interface by loading language packs to the IP phone. Language packs can only be loaded using configuration files.
The following table lists available languages and associated language packs.
Available Language
English
Chinese_S (only applicable to
SIP-T19P and SIP-T21P IP phones)
Associated Language Pack lang+English.txt lang-Chinese_S.txt
Chinese_T (only applicable to
SIP-T19P and SIP-T21P IP phones)
German
French
Italian
Portuguese
Polish
Spanish
Turkish lang-Chinese_T.txt lang-German.txt lang-French.txt lang-Italian.txt lang-Portuguese.txt lang-Polish.txt lang-Spanish.txt lang-Turkish.txt
Procedure
Loading language pack can only be performed using the configuration files.
Configuration File <y0000000000xx>.cfg
Specify the access URL of the language pack.
For more information, refer to
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
The default language used on the phone user interface is English. The default language used on the web user interface depends on the language preferences in the browser (if the language is not supported by the IP phone, the web user interface uses English). You can specify the languages for the phone user interface and web user interface respectively.
Procedure
Specify the language for the phone user interface or the web user interface using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Specify the languages for the phone user interface and the web user interface.
For more information, refer to
Specify the language for the web user interface.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-preference&q=load
Specify the language for the phone user interface.
To specify the language for the web user interface via web user interface:
1. Click on Settings->Preference.
2. Select the desired language from the pull-down list of Language.
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3. Click Confirm to accept the change.
Configuring Basic Features
To specify the language for the phone user interface via phone user interface:
1. Press Menu->Settings->Basic Settings->Language.
2. Press or to select the desired language.
3. Press the Save soft key to accept the change.
Logo customization allows unifying the IP phone appearance or displaying a custom image on the idle screen such as a company logo, instead of the default system logo.
SIP-T20P IP phones only support a text logo.
The following table lists the logo file format and resolution for each phone model.
Phone Model
SIP-T28P
SIP-T26P
SIP-T22P/T21P/T19P
Logo File Format
.dob
.dob
.dob
Resolution
<=236*82 2 gray scale
<=132*64 2 gray scale
<=132*64 2 gray scale
Note The format of the logo file must be *.dob. Before uploading your custom logo to IP phones, ensure your logo file is correctly formatted. For more information on customizing a logo file, refer to Yealink_SIP-T2
Series_T19P_T4_Series_IP_Phones_Auto_Provisioning_Guide .
Procedure
The logo shown on the idle screen can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the logo shown on the idle screen.
For more information, refer to
on page
Configure the logo shown on the idle screen.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
To configure an image logo via web user interface (not applicable to SIP-T20P IP phones):
1. Click on Features->General Information.
2. Select Custom logo from the pull-down list of Use Logo.
3. Click Browse to select the logo file from your local system.
4. Click Upload to upload the file.
5. Click Confirm to accept the change.
For SIP-T28P IP phones, the image logo is displayed on the idle screen. For
SIP-T26P/T22P IP phones, the image logo screen and the idle screen are displayed alternately.
To configure a text logo via web user interface (only applicable to SIP-T20P IP phones):
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of User Logo.
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Configuring Basic Features
3. Enter the desired text (0~15 characters) in the Text Logo field.
4. Click Confirm to accept the change.
The registered account and the configured text logo are displayed alternately.
Softkey layout is used to customize the soft keys at the bottom of the LCD screen to best meet users’ requirements. It can be configured based on call states. In addition to specifying which soft keys to display, you can determine their display order. Softkey layout is not applicable to SIP-T20P IP phones. You can create softkey layout templates for different call states. For more information on the softkey layout template, refer to
Softkey Layout Template on page 231 .
The following table lists soft keys available for IP phones in different call states.
CallFailed
CallIn
Call State Default Soft Keys
NewCall
Empty
Empty
Empty
Answer
Forward
Silence
Reject
Optional Soft Keys
Empty
Switch
Cancel
Empty
Switch
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Connecting
Dialing
RingBack
Talking
Call State
Connecting
SemiAttendTrans
RingBack
Empty
Empty
Empty
Cancel
SemiAttendTransBack
Transfer
Empty
Empty
Cancel
Talk
Transfer
Hold
Conference
Cancel
Hold
Default Soft Keys
Empty
Empty
Empty
Cancel
Transfer
Empty
Empty
Cancel
Send
IME
Delete
Cancel
Optional Soft Keys
Empty
Switch
Empty
Switch
Empty
History
Switch
Line
Favorite
GPickup
DPickup
Empty
Switch
CC
Transfer
Resume
NewCall
Cancel
Empty
Switch
CC
Empty
Mute
SWAP
NewCall
Switch
Answer
Reject
Empty
Switch
Answer
Reject
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Configuring Basic Features
Call State
Held
PreTrans
Conferenced
Default Soft Keys
Empty
Empty
Empty
Cancel
Transfer
IME
Delete
Cancel
Empty
Hold
Split
Cancel
Optional Soft Keys
Empty
Switch
Answer
Reject
NewCall
Empty
Directory
Switch
Send
Empty
Switch
Answer
Reject
Mute
Procedure
Softkey layout can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Specify the access URL of the softkey layout template.
For more information, refer to
Access URL of Softkey Layout on page 393 .
Configure the softkey layout.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-softkey&q=load
To configure softkey layout via web user interface:
1. Click on Settings->Softkey Layout.
2. Select the desired value from the pull-down list of Custom Softkey.
3. Select the desired state from the pull-down list of Call States.
4. Select the desired soft key from the Unselected Softkeys column and then click .
The selected soft key appears in the Selected Softkeys column.
5. Repeat the step 4 to add more soft keys to the Selected Softkeys column.
6. To remove the soft key from the Selected Softkeys column, select the desired soft
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones key and then click .
7. To adjust the display order of soft keys, select the desired soft key and then click or .
The LCD screen displays the soft keys in the adjusted order.
8. Click Confirm to accept the change.
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Key as send allows assigning the pound key or star key as a send key. Send sound allows the IP phone to play a key tone when a user presses the send key. Key tone allows the IP phone to play a key tone when a user presses any key. Send sound works only if Key tone is enabled.
Procedure
Key as send can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure a send key.
Configure a send sound.
Configure a key tone.
For more information, refer to Key as Send on page 290 .
Configure a send key.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure a send sound and key tone.
Navigate to:
Configuring Basic Features
Phone User Interface http://<phoneIPAddress>/servlet
?p=features-audio&q=load
Configure the send key.
To configure a send key via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Key As Send.
3. Click Confirm to accept the change.
To configure a send sound and key tone via web user interface:
1. Click on Features->Audio.
2. Select the desired value from the pull-down list of Key Sound.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
3. Select the desired value from the pull-down list of Send Sound.
4. Click Confirm to accept the change.
To configure send key via phone user interface:
1. Press Menu->Features->Key as Send.
2. Press or , or the Switch soft key to select # or * from the Key as Send field, or select Disable to disable this feature.
3. Press the Save soft key to accept the change.
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Hotline is a point-to-point communication link in which a call is automatically directed to the preset hotline number. The IP phone automatically dials out the hotline number using the first available line after a specified time interval when off-hook. IP phones only support one hotline number.
Procedure
Hotline can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the hotline number.
Specify the time (in seconds) the
IP phone waits before automatically dialing out the hotline number.
For more information, refer to
on page
Configure the hotline number.
Specify the time (in seconds) the
Configuring Basic Features
Phone User Interface
IP phone waits before automatically dial out the hotline number.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure the hotline number.
Specify the time (in seconds) the
IP phone waits before automatically dialing out the hotline number.
To configure hotline via web user interface:
1. Click on Features->General Information.
2. Enter the hotline number in the Hotline Number field.
3. Enter the delay time in the Hotline Delay (0~10s) field.
4. Click Confirm to accept the change.
To configure hotline via phone user interface:
1. Press Menu->Features->Hot Line.
2. Enter the hotline number in the Hot Number field.
3. Enter the waiting time (in seconds) in the HotLine Delay field.
4. Press the Save soft key to accept the change.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Call log contains call information such as remote party identification, time and date, and call duration. IP phones maintain a local call log. Call log consists of four lists:
Placed Calls, Received Calls, Missed Calls and Forwarded Calls. Call log lists support
100 entries in all. To store call information, you must enable save call log feature in advance.
Procedure
Call log can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure call log feature.
For more information, refer to Call
on page
Configure call log feature.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure the call log.
To configure call log feature via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Save Call Log.
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3. Click Confirm to accept the change.
Configuring Basic Features
To configure call log feature via phone user interface:
1. Press Menu->Features->History Setting.
2. Press or , or the Switch soft key to select the desired value from the History
Record field.
3. Press the Save soft key to accept the change.
Missed call log allows the IP phone to display the number of missed calls with an indicator icon on the idle screen, and to log missed calls in the Missed Calls list when the IP phone misses calls. It is configurable on a per-line basis. Once the user accesses the Missed Calls list, the prompt message and indicator icon on the idle screen disappear.
Procedure
Missed call log can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure missed call log feature.
For more information, refer to
Configure missed call log feature.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-basic&q=load&acc
=0
To configure missed call log via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Basic.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
4. Select the desired value from the pull-down list of Missed Call Log.
5. Click Confirm to accept the change.
IP phones maintain a local directory. The local directory can store up to 1000 contacts and 5 groups. When adding a contact to the local directory, in addition to name and phone numbers, you can also specify the account, ring tone and group for the contact.
Contacts and groups can be added either one by one or in batch using a local contact
file. For more information on the contact file, refer to Local Contact File on page 233 .
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Specify the access URL of the local contact file.
For more information, refer to
Access URL of Local Contact File on page 396 .
Add a group and a contact to the local directory.
Navigate to: http://<phoneIPAddress>/servlet
?p=contactsbasic&q=load&num
=1&group=
Add a group and a contact to the local directory.
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Configuring Basic Features
To add a group to the local directory via web user interface:
1. Click on Directory->Local Directory.
2. In the Group Setting block, enter the desired group name in the Group field.
3. Select the desired ring tone from the pull-down list of Ring field.
4. Click Add to add the group.
To add a contact to the local directory via web user interface:
1. Click on Directory->Local Directory.
2. In the Directory block, enter the name and the office, mobile or other numbers in the corresponding fields.
3. Select the desired ring tone from the pull-down list of Ring Tone.
4. Select the desired group from the pull-down list of Group.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
5. Select the desired account from the pull-down list of Account.
If Auto is selected, the IP phone will use the first available account when placing calls to the contact from the local directory.
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6. Click Add to add the contact.
To add a group to the local directory via phone user interface:
1. Press Menu->Directory->Local Directory.
2. Press the AddGrp soft key.
3. Enter the desired group name in the Name field.
4. Press or , or the Switch soft key to select the desired group ring tone from the Ring Tones field.
5. Press the Add soft key to accept the change.
To add a contact to the local directory via phone user interface:
1. Press Menu->Directory->Local Directory.
2. Select the desired contact group.
3. Press the Add soft key.
4. Enter the name and the office, mobile or other numbers in the corresponding fields.
5. Press or , or the Switch soft key to select the desired account from the
Account field.
If Auto is selected, the IP phone will use the first available account when placing calls to the contact from the local directory.
6. Press or , or the Switch soft key to select the desired ring tone from the Ring
Tones field.
7. Press the Save soft key to accept the change.
Configuring Basic Features
Live dialpad allows IP phones to automatically dial out the entered phone number after a specified period of time.
Procedure
Live dialpad can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure live dialpad.
For more information, refer to Live
Configure live dialpad.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-preference&q=load
To configure live dialpad via web user interface:
1. Click on Settings->Preference.
2. Select the desired value from the pull-down list of Live Dialpad.
3. Enter the desired delay time in the Inter Digit Time (1~14s) field.
4. Click Confirm to accept the change.
Call waiting allows IP phones to receive a new call when there is already an active call. The new incoming call is presented to the user visually on the LCD screen. Call waiting tone allows the phone to play a short tone, to remind the user audibly of a new incoming call during conversation. Call waiting tone works only if call waiting is
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones enabled.
Procedure
Call waiting and call waiting tone can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure call waiting and call waiting tone.
For more information, refer to Call
Configure call waiting.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure call waiting tone.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-audio&q=load
Configure call waiting and call waiting tone.
To configure call waiting via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Call Waiting.
3. (Optional.) Enter the call waiting on code in the Call Waiting On Code field.
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Configuring Basic Features
4. (Optional.) Enter the call waiting off code in the Call Waiting Off Code field.
5. Click Confirm to accept the change.
To configure call waiting tone via web user interface:
1. Click on Features->Audio.
2. Select the desired value from the pull-down list of Call Waiting Tone.
3. Click Confirm to accept the change.
To configure call waiting and call waiting tone via phone user interface:
1. Press Menu->Features->Call Waiting.
2. Press or , or the Switch soft key to select the desired value from the Call
Waiting field.
3. Press or , or the Switch soft key to select the desired value from the Play
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Tone field.
4. (Optional.) Enter the call waiting on code in the CW On Code field.
5. (Optional.) Enter the call waiting off code in the CW Off Code field.
6. Press the Save soft key to accept the change.
Auto redial allows IP phones to redial a busy number after the first attempt. Both the number of attempts and waiting time between redials are configurable.
Procedure
Auto redial can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure auto redial feature.
For more information, refer to
on page
Configure auto redial feature.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure auto redial feature.
To configure auto redial via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Auto Redial.
3. Enter the waiting time in the Auto Redial Interval (1~300s) field.
The default waiting time is 10s.
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Configuring Basic Features
4. Enter the desired times in the Auto Redial Times (1~300) field.
The default value is 10.
5. Click Confirm to accept the change.
To configure auto redial via phone user interface:
1. Press Menu->Features->Auto Redial.
2. Press or , or the Switch soft key to select the desired value from the Auto
Redial field.
3. Enter the waiting time (in seconds) in the Redial Interval field.
4. Enter the desired times in the Redial Times field.
5. Press the Save soft key to accept the change.
Auto answer allows IP phones to automatically answer an incoming call. IP phones will not automatically answer the incoming call during a call even if auto answer is enabled.
Auto answer is configurable on a per-line basis. Auto-Answer delay defines a period of delay time before the IP phone automatically answers incoming calls.
Procedure
Auto answer can be configured using the configuration files or locally.
Configuration File <MAC>.cfg
Configure auto answer.
For more information, refer to
on page
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Specify a period of delay time for auto answer.
For more information, refer to
on page
Configure auto answer.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-basic&q=load&acc
=0
Specify a period of delay time for auto answer.
Navigate to: http://<phoneIPAddress>servlet?
p=features-general&q=load
Configure auto answer.
To configure auto answer via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Basic.
4. Select the desired value from the pull-down list of Auto Answer.
5. Click Confirm to accept the change.
To configure a period of delay time for auto answer via web user interface:
1. Click on Features->General Information.
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Configuring Basic Features
2. Enter the desired time in the Auto-Answer Delay (1~4s) field.
3. Click Confirm to accept the change.
To configure auto answer via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Accounts.
2. Select the desired account and then press the Enter soft key.
3. Press or , or the Switch soft key to select the desired value from the Auto
Answer field.
4. Press the Save soft key to accept the change.
Call completion allows users to monitor the busy party and establish a call when the busy party becomes available to receive a call. Two factors commonly prevent a call from connecting successfully:
Callee does not answer
Callee actively rejects the incoming call before answering
IP phones support call completion using the SUBSCRIBE/NOTIFY method, which is specified in draft-poetzl-sipping-call-completion-00, to subscribe to the busy party and receive notifications of their status changes.
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Procedure
Call completion can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure call completion.
For more information, refer to Call
Configure call completion.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure call completion.
To configure call completion via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Call Completion.
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3. Click Confirm to accept the change.
To configure call completion via phone user interface:
1. Press Menu->Features->Call Completion.
2. Press or , or the Switch soft key to select the desired value from the Call
Completion field.
3. Press the Save soft key to accept the change.
Configuring Basic Features
Anonymous call allows the caller to conceal the identity information displayed on the callee’s screen. The callee’s phone LCD screen prompts an incoming call from anonymity. Anonymous call is configurable on a per-line basis.
Example of anonymous SIP header:
Via: SIP/2.0/UDP 10.2.8.183:5063;branch=z9hG4bK1535948896
From: "Anonymous" <sip:[email protected]>;tag=128043702
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5063>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER,
PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T28P 2.72.0.1
Privacy: id
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
P-Preferred-Identity: <sip:[email protected]>
Content-Length: 302
The anonymous call on code and anonymous call off code configured on IP phones are used to activate/deactivate the server-side anonymous call feature. They may vary on different servers. Send Anonymous Code feature allows IP phones to send anonymous on/off code to the server.
Procedure
Anonymous call can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Phone User Interface
Configure anonymous call.
For more information, refer to
Configure anonymous call.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-basic&q=load&acc
=0
Configure anonymous call.
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To configure anonymous call via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Basic.
4. Select the desired value from the pull-down list of Local Anonymous.
5. Select the desired value from the pull-down list of Send Anonymous Code.
6. (Optional.) Enter the anonymous call on code in the On Code field.
7. (Optional.) Enter the anonymous call off code in the Off Code field.
8. Click Confirm to accept the change.
To configure the anonymous call via phone user interface:
1. Press Menu->Features->Anonymous Call.
2. Press or , or the Switch soft key to select the desired line from the Line ID field.
3. Press or , or the Switch soft key to select the desired value from the Send
Anon field.
4. Press or , or the Switch soft key to select the desired value from the Anon
Code field.
5. (Optional.) Enter the anonymous call on code in the Call On Code field.
6. (Optional.) Enter the anonymous call off code in the Call Off Code field.
7. Press the Save soft key to accept the change.
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Anonymous call rejection allows IP phones to automatically reject incoming calls from callers whose identity has been deliberately concealed. The anonymous caller’s phone
LCD screen presents “Anonymity Disallowed”. Anonymous call rejection is configurable on a per-line basis.
Configuring Basic Features
The anonymous call rejection on code and anonymous call rejection off code configured on IP phones are used to activate/deactivate the server-side anonymous call rejection feature. They may vary on different servers.
Procedure
Anonymous call rejection can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Phone User Interface
Configure anonymous call rejection.
For more information, refer to
Anonymous Call Rejection on page 301 .
Configure anonymous call rejection.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-basic&q=load&acc
=0
Configure anonymous call rejection.
To configure anonymous call rejection via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Basic.
4. Select the desired value from the pull-down list of Anonymous Call Rejection.
5. (Optional.) Enter the anonymous call rejection on code in the On Code field.
6. (Optional.) Enter the anonymous call rejection off code in the Off Code field.
7. Click Confirm to accept the change.
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To configure anonymous call rejection via phone user interface:
1. Press Menu->Features->Anonymous Call.
2. Press or , or the Switch soft key to select the desired line from the Line ID field.
3. Press or , or the Switch soft key to select the desired value from the Anon
Rejection field.
4. (Optional.) Enter the anonymous call rejection on code in the Reject On Code field.
5. (Optional.) Enter the anonymous call rejection off code in the Reject Off Code field.
6. Press the Save soft key to accept the change.
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Do Not Disturb (DND) allows IP phones to ignore incoming calls. DND feature can be configured on a phone or a per-line basis depending on the DND mode. Two DND modes:
Phone (default): DND feature is effective for the IP phone.
Custom: DND feature can be configured for each or all accounts.
A user can activate or deactivate DND using the DND key or DND soft key (not applicable to SIP-T20P IP phones). DND activated on the IP phone disables the local call forward settings. The DND configurations on IP phones may be overridden by the server settings.
The DND on code and DND off code configured on IP phones are used to activate/deactivate the server-side DND feature. They may vary on different servers.
Return Message When DND
This feature defines the return code and the reason of the SIP response message for the rejected incoming call when DND is enabled on the IP phone. The caller’s phone LCD screen displays the received return code.
Procedure
DND can be configured using the configuration files or locally.
Configuration File
<MAC>.cfg
<y0000000000xx>.cfg
Configure DND in the custom mode.
For more information, refer to Do
on page
Assign a DND key.
For more information, refer to DND
on page
Configuring Basic Features
Local
Web User Interface
Phone User Interface
Configure the DND mode.
Configure DND in the phone mode.
Specify the return code and the reason of the SIP response message when DND is enabled.
For more information, refer to Do
Assign a DND key.
Navigate to: http://<phoneIPAddress>/servlet?
p=dsskey&q=load&model=0
Configure DND.
Navigate to: http://<phoneIPAddress>/servlet?
p=features-forward&q=load
Specify the return code and the reason of the SIP response message when DND is enabled.
Navigate to: http://<phoneIPAddress>/servlet?
p=features-general&q=load
Assign a DND key.
Configure DND.
To configure a DND key via web user interface:
1. Click on DSSKey->Memory Key (Line Key or Programable Key).
SIP-T19P IP phones only support programable keys and SIP-T22P/T21P/T20P IP phones only support line keys and programable keys.
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2. In the desired DSS key field, select DND from the pull-down list of Type.
3. Click Confirm to accept the change.
To configure DND feature via web user interface:
1. Click on Features->Forward & DND.
2. In the DND block, mark the desired radio box in the Mode field. a) If you mark the Phone radio box:
1) Mark the desired radio box in the DND Status field.
2) (Optional.) Enter the DND on code in the DND On Code field.
3) (Optional.) Enter the DND off code in the DND Off Code field.
86 b) If you mark the Custom radio box:
1) Select the desired account from the pull-down list of Account.
Configuring Basic Features
2) Mark the desired radio box in the DND Status field.
3) (Optional.) Enter the DND on code in the DND On Code field.
4) (Optional.) Enter the DND off code in the DND Off Code field.
3. Click Confirm to accept the change.
To specify the return code and the reason when DND is enabled via web user interface:
1. Click on Features->General Information.
2. Select the desired type from the pull-down list of Return Code When DND.
3. Click Confirm to accept the change.
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To configure a DND key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select DND from the Key Type field.
5. Press the Save soft key to accept the change.
To configure DND in the phone mode via phone user interface:
1. Press the DND soft key or the DND key when the IP phone is idle.
To configure DND in the custom mode for a specific account via phone user interface:
1. Press the DND soft key or the DND key when the IP phone is idle.
The LCD screen displays a list of accounts registered on the IP phone.
2. Press or to select the desired account.
3. Press or to select On to activate DND.
You can configure DND in the custom mode for all accounts by pressing the All On soft key.
4. Press the Save soft key to accept the change.
Busy tone is audible to the other party, indicating that the call connection has been broken when one party releases a call. Busy tone delay can define a period of time during which the busy tone is audible.
Procedure
Busy tone delay can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure busy tone delay.
For more information, refer to
on page
Configure busy tone delay.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
To configure busy tone delay via web user interface:
1． Click on Features->General Information.
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Configuring Basic Features
2． Select the desired value from the pull-down list of Busy Tone Delay (Seconds).
3． Click Confirm to accept the change.
Return code when refuse defines the return code and reason of the SIP response message for the refused call. The caller’s phone LCD screen displays the reason according to the received return code. Available return codes and reasons are:
404 (Not found)
480 (Temporarily not available)
486 (Busy here)
Procedure
Return code for refused call can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Specify the return code and the reason of the SIP response message when refusing a call.
For more information, refer to
page
Specify the return code and the reason of the SIP response message when refusing a call.
Navigate to:
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones http://<phoneIPAddress>/servlet
?p=features-general&q=load
To specify the return code and the reason when refusing a call via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Return Code When Refuse.
3. Click Confirm to accept the change.
Early media refers to media (e.g., audio and video) played to the caller before a SIP call is actually established. Current implementation supports early media through the
183 message. When the caller receives a 183 message with SDP before the call is established, a media channel is established. This channel is used to provide the early media stream for the caller.
180 ring workaround defines whether to deal with the 180 message received after the
183 message. When the caller receives a 183 message, it suppresses any local ringback tone and begins to play the media received. 180 ring workaround allows IP phones to resume and play the local ringback tone upon a subsequent 180 message received.
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Configuring Basic Features
Procedure
180 ring workaround can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure 180 ring workaround.
For more information, refer to 180
Configur 180 ring workaround.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
To configure 180 ring workaround via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of 180 Ring Workaround.
3. Click Confirm to accept the change.
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An outbound proxy server can receive all initiating request messages and route them to the designated destination. If the IP phone is configured to use an outbound proxy server within a dialog, all SIP request messages from the IP phone will be sent to the outbound proxy server forcefully.
Note To use this feature, make sure the outbound server has been correctly configured on the
IP phone.
Procedure
Use outbound proxy in dialog can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Specify whether to use outbound proxy in a dialog.
For more information, refer to Use
Outbound Proxy in Dialog on page 306 .
Specify whether to use outbound proxy in a dialog.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
To specify whether to use outbound proxy server in a dialog via web user interface:
1. Click on Features->General Information.
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Configuring Basic Features
2. Select the desired value from the pull-down list of Use Outbound Proxy In Dialog.
3. Click Confirm to accept the change.
SIP session timers T1, T2 and T4 are SIP transaction layer timers defined in RFC 3261.
Timer T1 is an estimate of the Round Trip Time (RTT) of transactions between a SIP client and SIP server. Timer T2 represents the maximum retransmit interval for non-INVITE requests and INVITE responses. Timer T4 represents the maximum duration a message will remain in the network. These session timers are configurable on IP phones.
Procedure
SIP session timer can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure SIP session timer.
For more information, refer to SIP
Configure SIP session timer.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
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To configure session timer via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
4. Enter the desired value in the SIP Session Timer T1 (0.5~10s) field.
The default value is 0.5s.
5. Enter the desired value in the SIP Session Timer T2 (2~40s) field.
The default value is 4s.
6. Enter the desired value in the SIP Session Timer T4 (2.5~60s) field.
The default value is 5s.
7. Click Confirm to accept the change.
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Session timer allows a periodic refresh of SIP sessions through a re-INVITE request, to determine whether a SIP session is still active. Session timer is specified in RFC 4028. IP phones support two refresher modes: UAC and UAS. The UAC mode means refreshing the session from the client, while the UAS mode means refreshing the session from the server. The session expiration and session refresher are negotiated via the
Session-Expires header in the INVITE message. The negotiated refresher will send a re-INVITE/UPDATE request at or before the negotiated session expiration.
Configuring Basic Features
Procedure
Session timer can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure session timer.
For more information, refer to
Configure session timer.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
To configure session timer via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
4. Select the desired value from the pull-down list of Session Timer.
5. Enter the desired time interval in the Session Expires (30~7200s) field.
6. Select the desired refresher from the pull-down list of Session Refresher.
7. Click Confirm to accept the change.
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Call hold provides a service of placing an active call on hold. When a call is placed on hold, the IP phone sends an INVITE request with a HOLD SDP to the server. IP phones support two call hold methods, one is RFC 3264, which sets the “a” (media attribute) in the SDP to sendonly, recvonly or inactive (e.g., a=sendonly). The other is RFC 2543, which sets the “c” (connection addresses for the media streams) in the SDP to zero (e.g., c=0.0.0.0). Call hold tone allows IP phones to play a hold tone at regular intervals when there is a call on hold.
Procedure
Call hold can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the call hold tone and call hold tone delay.
Specify whether RFC 2543
(c=0.0.0.0) outgoing hold signaling is used.
For more information, refer to Call
Configure the call hold tone and call hold tone delay.
Specify whether RFC 2543
(c=0.0.0.0) outgoing hold signaling is used.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
To configure call hold method via web user interface:
1. Click on Features->General Information.
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Configuring Basic Features
2. Select the desired value from the pull-down list of RFC 2543 Hold.
3. Click Confirm to accept the change.
To configure call hold tone and call hold tone delay via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Play Hold Tone.
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3. Enter the desired time in the Play Hold Tone Delay field.
4. Click Confirm to accept the change.
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Call forward allows users to redirect an incoming call to a third party. IP phones redirect an incoming INVITE message by responding with a 302 Moved Temporarily message, which contains a Contact header with a new URI that should be tried. Three types of call forward:
Always Forward -- Forward the incoming call immediately.
Busy Forward -- Forward the incoming call when the callee is busy.
No Answer Forward -- Forward the incoming call after a period of ring time.
Call forward can be configured on a phone or a per-line basis depending on the call forward mode. The following describes the call forward modes:
Phone (default): Call forward feature is effective for the IP phone.
Custom: Call forward feature can be configured for each or all accounts.
The call forward on code and call forward off code configured on IP phones are used to activate/deactivate the server-side call forward feature. They may vary on different servers.
Forward International
Forward international allows users to forward an incoming call to an international telephone number. This feature is enabled by default.
Configuring Basic Features
Procedure
Call forward can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure call forward in custom mode.
For more information, refer to
on page
Configure the call forward mode.
Configure call forward in phone mode.
Configure forward international.
For more information, refer to
on page
Configure call forward.
Navigate to: http://<phoneIPAddress>/ser vlet?p=features-forward&q=l oad
Configure forward international.
Navigate to: http://<phoneIPAddress>/ servlet?p=features-general& q=load
Configure call forward.
To configure call forward via web user interface:
1. Click on Features->Forward & DND.
2. In the Forward block, mark the desired radio box in the Mode field. a) If you mark the Phone radio box:
1) Mark the desired radio box in the Always/Busy/No Answer Forward field.
2) Enter the destination number you want to forward in the Target field.
3) (Optional.) Enter the on code and off code in the On Code and Off Code fields.
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4) Select the ring time to wait before forwarding from the pull-down list of After
Ring Time (0~120s) (only for the no answer forward). b) If you mark the Custom radio box:
1) Select the desired account from the pull-down list of Account.
2) Mark the desired radio box in the Always/Busy/No Answer Forward field.
3) Enter the destination number you want to forward in the Target field.
4) Enter the on code and off code in the On Code and Off Code fields.
5) Select the ring time to wait before forwarding from the pull-down list of After
Ring Time (0~120s) (only for the no answer forward).
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3. Click Confirm to accept the change.
To configure forward international via web user interface:
1. Click on Features->General Information.
Configuring Basic Features
2. Select the desired value from the pull-down list of Fwd International.
3. Click Confirm to accept the change.
To configure call forward in phone mode via phone user interface:
1. Press Menu->Features->Call Forward.
2. Press or to select the desired forwarding type, and then press the Enter soft key.
3. Depending on your selection: a) If you select Always Forward:
1) Press or , or the Switch soft key to select the desired value from the
Always field.
2) Enter the destination number you want to forward all incoming calls to in the
Forward To field.
3) (Optional.) Enter the always forward on code and off code respectively in the
On Code and Off Code fields. b) If you select Busy Forward:
1) Press or , or the Switch soft key to select the desired value from the
Busy field.
2) Enter the destination number you want to forward all incoming calls to when the IP phone is busy in the Forward To field.
3) (Optional.) Enter the busy forward on code and off code respectively in the
On Code and Off Code fields. c) If you select No Answer Forward:
1) Press or , or the Switch soft key to select the desired value from the
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No Answer field.
2) Enter the destination number you want to forward all unanswered incoming calls to in the Forward To field.
3) Press or , or the Switch soft key to select the ring time to wait before forwarding from the After Ring Time field.
The default ring time is 12 seconds.
4) (Optional.) Enter the no answer forward on code and off code respectively in the On Code and Off Code fields.
4. Press the Save soft key to accept the change.
To configure call forward in custom mode via phone user interface:
1. Press Menu->Features->Call Forward.
2. Press or to select the desired account, and then press the Enter soft key.
3. Press or to select the desired forwarding type, and then press the Enter soft key.
4. Depending on your selection: a) If you select Always Forward, you can configure it for a specific account.
1) Press or , or the Switch soft key to select the desired value from the
Always field.
2) Enter the destination number you want to forward all incoming calls to in the
Forward To field.
3) (Optional.) Enter the always forward on code and off code respectively in the
On Code and Off Code fields.
You can also configure the always forward for all accounts. After the always forward was configured for a specific account, do the following:
1) Press or to highlight the Always field.
2) Press the All Lines soft key.
The LCD screen prompts “Copy to All Lines?”.
3) Press the OK soft key to accept the change. b) If you select Busy Forward, you can configure it for a specific account.
1) Press or , or the Switch soft key to select the desired value from the
Busy field.
2) Enter the destination number you want to forward all incoming calls to when the IP phone is busy in the Forward To field.
3) (Optional.) Enter the busy forward on code and off code respectively in the
On Code and Off Code fields.
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Configuring Basic Features
You can also configure the busy forward for all accounts. After the busy forward was configured for a specific account, do the following:
1) Press or to highlight the Busy field.
2) Press the All Lines soft key.
The LCD screen prompts “Copy to All Lines?”.
3) Press the OK soft key to accept the change. c) If you select No Answer Forward, you can configure it for a specific account.
1) Press or , or the Switch soft key to select the desired value from the
No Answer field.
2) Enter the destination number you want to forward all unanswered incoming calls to in the Forward To field.
3) Press or , or the Switch soft key to select the ring time to wait before forwarding from the After Ring Time field
The default ring time is 12 seconds.
4) (Optional.) Enter the no answer forward on code and off code respectively in the On Code and Off Code fields.
You can also configure the no answer forward for all accounts. After the no answer forward was configured for a specific account, do the following:
1) Press or to highlight the No Answer field.
2) Press the All Lines soft key.
The LCD screen prompts “Copy to All Lines?”.
3) Press the OK soft key to accept the change.
5. Press the Save soft key to accept the change.
Call transfer enables IP phones to transfer an existing call to another party. IP phones support call transfer using the REFER method specified in RFC 3515 and offer three types of transfer:
Blind Transfer -- Transfer a call directly to another party without consulting. Blind transfer is implemented by a simple REFER method without Replaces in the Refer-To header.
Semi-attended Transfer -- Transfer a call after hearing the ringback tone.
Semi-attended transfer is implemented by a REFER method with Replaces in the
Refer-To header.
Attended Transfer -- Transfer a call with prior consulting. Attended transfer is implemented by a REFER method with Replaces in the Refer-To header.
Normally, call transfer is completed by pressing the transfer key. Blind transfer on hook and semi-attended transfer on hook features allow the IP phone to complete the
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones transfer through on-hook.
When a user performs a semi-attended transfer, semi-attended transfer feature determines whether to display the prompt "n New Missed Call(s)" ("n" indicates the number of the missed calls) on the destination party’s phone LCD screen.
Procedure
Call transfer can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Specify whether to complete the transfer through on-hook.
Configure semi-attended transfer feature.
For more information, refer to Call
Specify whether to complete the transfer through on-hook.
Configure semi-attended transfer feature.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-transfer&q=load
To configure call transfer via web user interface:
1. Click on Features->Transfer.
2. Select the desired values from the pull-down lists of Semi-Attend Transfer, Blind
Transfer On Hook and Semi Attend Transfer On Hook.
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3. Click Confirm to accept the change.
Configuring Basic Features
Network conference, also known as centralized conference, provides users with flexibility of call with multiple participants (more than three). IP phones implement network conference using the REFER method specified in RFC 4579. This feature depends on support from a SIP server.
Procedure
Network conference can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure network conference.
For more information, refer to
Configure network conference.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
To configure the network conference via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
4. Select Network Conference from the pull-down list of Conference Type.
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5. Enter the conference URI in the Conference URI field.
6. Click Confirm to accept the change.
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For local conference, all parties drop the call when the conference initiator drops the conference call. Transfer on conference hang up allows the other two parties to remain connected when the conference initiator drops the conference call.
Procedure
Transfer on conference hang up can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the transfer on conference hang up.
For more information, refer to
Transfer on Conference Hang Up
on page
Configure the transfer on conference hang up.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-transfer&q=load
Configuring Basic Features
To configure Transfer on Conference Hang up via web user interface:
1. Click on Features->Transfer.
2. Select the desired value from the pull-down list of Transfer on Conference Hang up.
3. Click Confirm to accept the change.
Directed call pickup is used for picking up an incoming call on a specific extension. A user can pick up the incoming call using a directed pickup key or the DPickup soft key
(not applicable to SIP-T20P IP phones). This feature depends on support from a SIP server.
For many SIP servers, directed call pickup requires a directed pickup code, which can be configured on a phone or a per-line basis.
Note It is recommended not to configure the directed call pickup key and the DPickup soft key simultaneously. If you do, the directed call pickup key will not be used correctly.
Procedure
Directed call pickup can be configured using the configuration files or locally.
Configuration File <MAC>.cfg
Configure the directed call pickup code on a per-line basis.
Configure directed call pickup feature on a phone basis.
For more information, refer to
on page
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Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Assign a directed call pickup key.
For more information, refer to
on page
Assign a directed call pickup key.
Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&q=load&model=
0
Configure directed call pickup feature on a phone basis.
Navigate to: http://<phoneIPAddress>/servl et?p=features-callpickup&q=lo ad
Configure directed call pickup code on a per-line basis.
Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac c=0
Assign a directed call pickup key.
To configure a directed call pickup key via web user interface:
1. Click on DSSKey->Memory Key (Line Key or Programable Key).
SIP-T19P IP phones only support programable keys and SIP-T22P/T21P/T20P IP phones only support line keys and programable keys.
2. In the desired DSS key field, select Directed Pickup from the pull-down list of Type.
3. Enter the directed call pickup code followed by the specific extension in the Value field.
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Configuring Basic Features
4. Select the desired line from the pull-down list of Line.
5. Click Confirm to accept the change.
To configure directed call pickup feature on a phone basis via web user interface:
1. Click on Features->Call Pickup.
2. Select the desired value from the pull-down list of Directed Call Pickup (not applicable to SIP-T20P IP phones).
3. Enter the directed call pickup code in the Directed Call Pickup Code field.
4. Click Confirm to accept the change.
To configure the directed call pickup code on a per-line basis via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
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4. Enter the directed call pickup code in the Directed Call Pickup Code field.
5. Click Confirm to accept the change.
To configure a directed pickup key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Directed Pickup from the Key Type field.
5. Press or , or the Switch soft key to select the desired line from the Account
ID field.
6. Enter the directed call pickup code followed by the specific extension in the Value field.
7. Press the Save soft key to accept the change.
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Group call pickup is used for picking up incoming calls within a pre-defined group. If the group receives many incoming calls at once, the user will pick up the first incoming call, using a group pickup key or the GPickup soft key (not applicable to SIP-T20P IP phones).
This feature depends on support from a SIP server. For many SIP servers, group call pickup requires a group pickup code, which can be configured on a phone or a per-line basis.
Configuring Basic Features
Procedure
Group call pickup can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure the group call pickup code on a per-line basis.
Configure group call pickup feature on a phone basis.
For more information, refer to
Group Call Pickup on page 324 .
Assign a group call pickup key.
For more information, refer to
on page
Assign a group call pickup key.
Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&q=load&model=
0
Configure group call pickup feature on a phone basis.
Navigate to: http://<phoneIPAddress>/servl et?p=features-callpickup&q=lo ad
Configure the group call pickup code on a per-line basis.
Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac c=0
Assign a group call pickup key.
To configure a group call pickup key via web user interface:
1. Click on DSSKey->Memory Key (Line Key or Programable Key).
SIP-T19P IP phones only support programable keys and SIP-T22P/T21P/T20P IP phones only support line keys and programable keys.
2. In the desired DSS key field, select Group Pickup from the pull-down list of Type.
3. Enter the group call pickup code in the Value field.
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4. Select the desired line from the pull-down list of Line.
5. Click Confirm to accept the change.
To configure group call pickup feature on a phone basis via web user interface:
1. Click on Features->Call Pickup.
2. Select the desired value from the pull-down list of Group Call Pickup (not applicable to SIP-T20P IP phones).
3. Enter the group call pickup code in the Group Call Pickup Code field.
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4. Click Confirm to accept the change.
To configure the group call pickup code on a per-line basis via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
Configuring Basic Features
4. Enter the group call pickup code in the Group Call Pickup Code field.
5. Click Confirm to accept the change.
To configure a group pickup key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Group Pickup from the Key Type field.
5. Press or , or the Switch soft key to select the desired line from the Account
ID field.
6. Enter the group call pickup code in the Value field.
7. Press the Save soft key to accept the change.
Call pickup is implemented through SIP signals on some specific servers. IP phones support to pick up incoming calls via a NOTIFY message with dialog-info event. A user can pick up an incoming call by pressing the DSS key used to monitor a specific extension (such as the BLF key). This feature is not applicable to SIP-T19P IP phones.
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Example of the dialog-info message carried in NOTIFY message:
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="6" state="full" entity="sip:[email protected]">
<dialog id="[email protected]" call-id="[email protected]" local-tag="827932784" remote-tag="1887460740" direction="recipient">
<state>early</state>
<local>
<identity>sip:[email protected]</identity>
<target uri="sip:[email protected]">
</target>
</local>
<remote>
<identity>sip:[email protected]</identity>
<target uri="sip:[email protected]:5063">
</target>
</remote>
</dialog>
</dialog-info>
Procedure
Dialog info call pickup can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure dialog info call pickup.
For more information, refer to
Dialog Info Call Pickup on page
Configure dialog info call pickup.
Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac c=0
To configure dialog info call pickup via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
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Configuring Basic Features
4. Select the desired value from the pull-down list of Dialog Info Call Pickup.
5. Click Confirm to accept the change.
Call return, also known as last call return, allows users to place a call back to the last caller. Call return is implemented on IP phones using a call return key.
Procedure
Call return key can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Assign a call return key.
For more information, refer to Call
on page
Assign a call return key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&q=load&model=0
Assign a call return key.
To configure a call return key via web user interface:
1. Click on DSSKey->Memory Key (Line Key or Programable Key).
SIP-T19P IP phones only support programable keys and SIP-T22P/T21P/T20P IP phones only support line keys and programable keys.
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2. In the desired DSS key field, select Call Return from the pull-down list of Type.
3. Click Confirm to accept the change.
To configure a call return key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Call Return from the Key Type field.
5. Press the Save soft key to accept the change.
Call park allows users to park a call on a special extension and then retrieve it on any other phone in the system. Users can park calls on the extension, known as call park orbit, by pressing a call park key. The current call is placed on hold and can be retrieved on another IP phone. This feature depends on support from a SIP server.
Note SIP-T19P IP phones support call park feature for BroadSoft server only. For more information, refer to Yealink IP Phones Deployment Guide for BroadSoft UC-One
Environments .
Procedure
Call park key can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Assign a call park key.
For more information, refer to
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Configuring Basic Features
Local
Web User Interface
Phone User Interface
Assign a call park key.
Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&q=load&model=
0
Assign a call park key.
To configure a call park key via web user interface:
1. Click on DSSKey->Memory Key (or Line Key).
2. In the desired DSS key field, select Call Park from the pull-down list of Type.
3. Enter the desired value (e.g., call park feature code) in the Value field.
4. Select the desired line from the pull-down list of Line.
5. Click Confirm to accept the change.
To configure a call park key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Call Park from the Key Type field.
5. Press or , or the Switch soft key to select the desired line from the Account
ID field.
6. Enter the desired value (e.g., call park feature code) in the Value field.
7. Press the Save soft key to accept the change.
Web server type determines access protocol of the IP phone’s web user interface. IP phones support both HTTP and HTTPS protocols for accessing the web user interface.
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HTTP is an application protocol that runs on top of the TCP/IP suite of protocols. HTTPS is a web protocol that encrypts and decrypts user page requests as well as pages returned by the web server. Both HTTP and HTTPS port numbers are configurable.
Procedure
Web server type can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure the web access type,
HTTP port and HTTPS port.
For more information, refer to
Configure the web access type,
HTTP port and HTTPS port.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
Configure the web access type,
HTTP port and HTTPS port.
To configure web server type via web user interface:
1. Click on Network->Advanced.
2. Select the desired value from the pull-down list of HTTP.
3. Enter the HTTP port number in the HTTP Port (1~65535) field.
The default HTTP port number is 80.
4. Select the desired value from the pull-down list of HTTPS.
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Configuring Basic Features
5. Enter the HTTPS port number in the HTTPS Port (1~65535) field.
The default HTTPS port number is 443.
6. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
7. Click OK to reboot the IP phone.
To configure web server type via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin)
->Network->Webserver Type.
2. Press or , or the Switch soft key to select the desired value from the HTTP
Status field.
3. Enter the HTTP port number in the HTTP Port field.
4. Press or , or the Switch soft key to select the desired value from the HTTPS
Status field.
5. Enter the HTTPS port number in the HTTPS Port field.
6. Press the Save soft key to accept the change.
The IP phone reboots automatically to make settings effective after a period of time.
Calling line identification presentation (CLIP) allows IP phones to display the caller identity, derived from a SIP header contained in the INVITE message when receiving an incoming call. IP phones support deriving caller identity from three types of SIP header:
From, P-Asserted-Identity and Remote-Party-ID. Identity presentation is based on the
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones identity in the relevant SIP header.
If the caller has existed in the local directory, the local name assigned to the caller should be preferentially displayed.
Procedure
CLIP can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure the presentation of the caller identity.
For more information, refer to
Configure the presentation of the caller identity.
Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac c=0
To configure the presentation of the caller identity via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
4. Select the desired value from the pull-down list of the Caller ID Source.
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Configuring Basic Features
5. Click Confirm to accept the change.
Connected line identification presentation (COLP) allows IP phones to display the identity of the callee specified for outgoing calls. IP phones can display the Dialed Digits, or the identity in a SIP header (Remote-Party-ID or P-Asserted-Identity) received, or the identity in the From header carried in the UPDATE message sent by the callee as described in RFC 4916.
If the callee has existed in the directory, the local name assigned to the callee should be preferentially displayed.
Procedure
COLP can be configured only using the configuration files.
Configuration File <MAC>.cfg
Configure the presentation of the callee’s identity.
For more information, refer to
DTMF (Dual Tone Multi-frequency), better known as touch-tone, is used for telecommunication signaling over analog telephone lines in the voice-frequency band.
DTMF is the signal sent from the IP phone to the network, which is generated when pressing the IP phone’s keypad during a call. Each key pressed on the IP phone generates one sinusoidal tone of two frequencies. One is generated from a high frequency group and the other from a low frequency group.
The DTMF keypad is laid out in a 4× 4 matrix, with each row representing a low frequency, and each column representing a high frequency. Pressing a digit key (such as '1') will generate a sinusoidal tone for each of two frequencies (697 and 1209 hertz
(Hz)).
DTMF Keypad Frequencies:
697 Hz
770 Hz
852 Hz
941 Hz
1209 Hz
1
4
7
*
1336 Hz
2
5
8
0
1447 Hz
3
6
9
#
1633 Hz
A
B
C
D
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Three methods of transmitting DTMF digits on SIP calls:
RFC 2833 -- DTMF digits are transmitted by RTP Events compliant to RFC 2833.
INBAND -- DTMF digits are transmitted in the voice band.
SIP INFO -- DTMF digits are transmitted by SIP INFO messages.
The method of transmitting DTMF digits is configurable on a per-line basis.
RFC 2833
DTMF digits are transmitted using the RTP Event packets that are sent along with the voice path. These packets use RFC 2833 format and must have a payload type that matches what the other end is listening for. The payload type for RTP Event packets is configurable. IP phones default to 101 for the payload type, which use the definition to negotiate with the other end during call establishment.
The RTP Event packet contains 4 bytes. The 4 bytes are distributed over several fields denoted as Event, End bit, R-bit, Volume and Duration. If the End bit is set to 1, the packet contains the end of the DTMF event. You can configure the sending times of the end RTP Event packet.
INBAND
DTMF digits are transmitted within the audio of the IP phone conversation. It uses the same codec as your voice and is audible to conversation partners.
SIP INFO
DTMF digits are transmitted by the SIP INFO messages when the voice stream is established after a successful SIP 200 OK-ACK message sequence. The SIP INFO message is sent along the signaling path of the call. The SIP INFO message can support transmitting DTMF digits in three ways: DTMF, DTMF-Relay and Telephone-Event.
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
<MAC>.cfg
<y0000000000xx>.cfg
Configure the method of transmitting DTMF digit and the payload type.
For more information, refer to
Configure the number of times for the IP phone to send the end
RTP Event packet.
For more information, refer to
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Configuring Basic Features
Local Web User Interface
Configure the method of transmitting DTMF digits and the payload type.
Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac c=0
Configure the number of times for the IP phone to send the end
RTP Event packet.
Navigate to: http://<phoneIPAddress>/servl et?p=features-general&q=loa d
To configure the method of transmitting DTMF digits via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
4. Select the desired value from the pull-down list of DTMF Type.
5. If SIP INFO or AUTO or SIP INFO is selected, select the desired value from the pull-down list of DTMF Info Type.
6. Enter the desired value in the DTMF Payload Type (96~127) field.
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7. Click Confirm to accept the change.
To configure the number of times to send the end RTP Event packet via web user interface:
1. Click on Features->General Information.
2. Select the desired value (1-3) from the pull-down list of DTMF Repetition.
3. Click Confirm to accept the change.
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Suppress DTMF display allows IP phones to suppress the display of DTMF digits. DTMF digits are displayed as “*” on the LCD screen. Suppress DTMF display delay defines whether to display the DTMF digits for a short period of time before displaying as “*”.
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure suppress DTMF display and suppress DTMF display delay.
For more information, refer to
page
Configure suppress DTMF display and suppress DTMF
Configuring Basic Features display delay.
Navigate to: http://<phoneIPAddress>/servl et?p=features-general&q=loa d
To configure suppress DTMF display and suppress DTMF display delay via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Suppress DTMF Display.
3. Select the desired value from the pull-down list of Suppress DTMF Display Delay
(not applicable to SIP-T20P IP phones).
4. Click Confirm to accept the change.
Call transfer is implemented via DTMF on some traditional servers. The IP phone sends specified DTMF digits to the server for transferring calls to third parties.
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Configure transfer via DTMF.
For more information, refer to
Transfer via DTMF on page 331 .
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Local Web User Interface
Configure transfer via DTMF.
Navigate to: http://<phoneIPAddress>/servl et?p=features-general&q=loa d
To configure transfer via DTMF via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of DTMF Replace Tran.
3. Enter the specified DTMF digits in the Tran Send DTMF field.
4. Click Confirm to accept the change.
Intercom allows establishing an audio conversation directly. The IP phone can answer intercom calls automatically. This feature depends on support from a SIP server.
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Intercom is a useful feature in office environments to quickly connect with an operator or secretary. Users can press an intercom key to automatically initiate an outgoing intercom call with a remote extension. This feature is not applicable to SIP-T19P IP phones.
Configuring Basic Features
Procedure
Intercom key can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Assign an intercom key.
For more information, refer to
on page
Assign an intercom key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&q=load&model=0
Assign an intercom key.
To configure an intercom key via web user interface:
1. Click on DSSKey->Memory Key (or Line Key).
2. In the desired DSS key field, select Intercom from the pull-down list of Type.
3. Enter the remote extension number in the Value field.
4. Select the desired line from the pull-down list of Line.
5. Click Confirm to accept the change.
To configure an intercom key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Intercom from the Type field.
4. Select the desired line from the Account ID field.
5. Enter the remote extension number in the Value field.
6. Press the Save soft key to accept the change.
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The IP phone can process incoming calls differently depending on settings. There are four configuration options for incoming intercom calls:
Accept Intercom
Accept Intercom allows the IP phone to automatically answer an incoming intercom call.
Intercom Mute
Intercom Mute allows the IP phone to mute the microphone for incoming intercom calls.
Intercom Tone
Intercom Tone allows the IP phone to play a warning tone before answering an intercom call.
Intercom Barge
Intercom Barge allows the IP phone to automatically answer an incoming intercom call while an active call is in progress. The active call will be placed on hold.
Procedure
Incoming intercom calls can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure incoming intercom call feature.
For more information, refer to
on page
Configure incoming intercom call feature.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-intercom&q=load
Configure incoming intercom call feature.
To configure intercom via web user interface:
1. Click on Features->Intercom.
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Configuring Basic Features
2. Select the desired values from the pull-down lists of Accept Intercom, Intercom
Mute, Intercom Tone and Intercom Barge.
3. Click Confirm to accept the change.
To configure intercom via phone user interface:
1. Press Menu->Features->Intercom.
2. Press or , or the Switch soft key to select the desired values from the
Accept Intercom, Intercom Mute, Intercom Tone and Intercom Barge fields.
3. Press the Save soft key to accept the change.
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Configuring Advanced Features
This chapter provides information for making configuration changes for the following advanced features:
Distinctive ring tones allows certain incoming calls to trigger IP phones to play distinctive ring tones. The IP phone inspects the INVITE request for an "Alert-Info" header when receiving an incoming call. If the INVITE request contains an "Alert-Info" header, the IP phone strips out the URL and keyword parameter and maps them to the appropriate ring tone.
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Alert-Info headers in the following two formats:
Alert-Info: http://localIP/Bellcore-drN
Alert-Info: <URL>;info=info text;x-line-id=0
If the Alter-Info header contains the keyword “Bellcore-drN”, the IP phone will play the Bellcore-drN ring tone (N=1, 2, 3, 4 or 5) (if the parameter
“features.alert_info_tone” is set to 1).
Example:
Alert-Info: http://127.0.0.1/Bellcore-dr1
The following table identifies the different Bellcore ring tone patterns and cadences (These ring tones are designed for the BroadWorks server).
Bellcore
Tone
Bellcore-dr1
(standard)
Bellcore-dr2
Bellcore-dr3
Bellcore-dr4
Bellcore-dr5
Pattern
ID
Pattern Cadence
Ringing 2s On
Minimum
Duration
(ms)
1800
Nominal
Duration
(ms)
2000
Maximum
Duration
(ms)
2200
1
Silent
Ringing
4s Off
Long
3600
630
4000
800
4400
1025
2
Silent
Ringing Long
315
630
400
800
525
1025
3
4
5
Silent
Ringing
Silent
Ringing
Silent
Ringing
Silent
Ringing
Silent
Ringing
Silent
Ringing
Silent
Ringing
Short
Short
Long
Short
Long
Short
3475
315
145
315
145
630
2975
200
145
800
145
200
2975
450
4000
400
200
400
200
800
4000
300
200
1000
200
300
4000
500
525
525
1100
525
525
4400
550
4400
525
525
525
525
1025
4400
Note “Bellcore-dr5” is a ring splash tone that reminds the user that the DND or Always Call
Forward feature is enabled on the server side.
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Configuring Advanced Features
If the Alert-Info header contains a remote URL, the IP phone will try to download the
WAV ring tone file from the URL and then play the remote ring tone (if the parameter “account.X.alert_info_url_enable” is set to 1). If it fails to download the file, the IP phone will play the local ring tone associated with info text. If there is no text matched, the IP phone will play the preconfigured local ring tone in about ten seconds.
Example:
Alert-Info: http:<//192.168.0.12:8080/ring.wav>/info=family;x-line-id=0
Procedure
Distinctive ring tones can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Configure distinctive ring tones.
For more information, refer to
Distinctive Ring Tones on page
Configure the internal ringer text and internal ringer file.
For more information, refer to
Distinctive Ring Tones on page
Configure distinctive ring tones.
Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac c=0
Configure the internal ringer text and internal ringer file.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-ring&q=load
To configure distinctive ring tones via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
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4. Select the desired value from the pull-down list of Distinctive Ring Tones.
5. Click Confirm to accept the change.
To configure the internal ringer text and internal ringer file via web user interface:
1. Click on Settings->Ring.
2. Enter the keywords in the Internal Ringer Text fields.
3. Select the desired ring tones for each text from the pull-down lists of Internal Ringer
File.
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4. Click Confirm to accept the change.
When receiving a message, the IP phone will play a warning tone. You can customize tones or select specialized tone sets (vary from country to country) to indicate different conditions of the IP phone. The default tones used on IP phones are the US tone sets.
Available tone sets for IP phones:
Lithuania
India
Italy
Japan
Mexico
New Zealand
Netherlands
Norway
Portugal
Spain
Switzerland
Sweden
Russia
United States
Australia
Austria
Brazil
Belgium
China
Czech
Denmark
Finland
France
Germany
Great Britain
Greece
Hungary
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Chile
Czech ETSI
Configured tones can be heard on IP phones for the following conditions.
Condition
Dial
Ring Back
Busy
Congestion
Call Waiting
Dial Recall
Info
Stutter
Message
Description
When in the pre-dialing interface
Ring-back tone
When the callee is busy
When the network is congested
Call waiting tone
When receiving a call back
When receiving a special message
When receiving a voice mail
When receiving a text message
Note: It is not applicable to SIP-T20P IP phones.
When automatically answering a call Auto Answer
Procedure
Tones can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the tones for the IP phone.
For more information, refer to
Configure the tones for the IP phone.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-tones&q=load
To configure tones via web user interface:
1. Click on Settings->Tones.
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2. Select the desired type from the pull-down list of Select Country.
If you select Custom, you can customize a tone for each condition of the IP phone.
3. Click Confirm to accept the change.
Remote phone book is a centrally maintained phone book, stored on the remote server.
Users only need the access URL of the remote phone book. The IP phone can establish a connection with the remote server and download the phone book, and then display the remote phone book entries on the phone user interface. IP phones support up to 5 remote phone books. SIP-T28/T26P/T22P IP phones support up to 2500 remote phone book entries. SIP-T21P/T19P IP phones support up to 2000 remote phone book entries.
Remote phone book is customizable. For more information, refer to Remote XML Phone
Search Remote Phonebook Name allows IP phones to search the entry names from the remote phone book when receiving incoming calls. Search Flash Time defines how often IP phones refresh the local cache of the remote phone book.
Note Remote phone book is not applicable to SIP-T20P IP phones.
Procedure
Remote phone book can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Specify the access URL of the remote phone book.
Specify whether to query the entry name from the remote phone book when the IP phone
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Local Web User Interface receives an incoming call.
Specify how often the IP phone refreshes the local cache of the remote phone book.
For more information, refer to
on page
Specify the access URL of the remote phone book.
Navigate to: http://<phoneIPAddress>/servl et?p=contacts-remote&q=load
Specify whether to query the entry name from the remote phone book when the IP phone receives an incoming call.
Specify how often the IP phone refreshes the local cache of the remote phone book.
Navigate to: http://<phoneIPAddress>/servl et?p=contacts-remote&q=load
To specify access URL of the remote phone book via web user interface:
1. Click on Directory->Remote Phone Book.
2. Enter the access URL in the Remote URL field.
3. Enter the name in the Display Name field.
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4. Click Confirm to accept the change
Configuring Advanced Features
To configure Search Remote Phonebook Name and Search Flash Time via web user interface:
1. Click on Directory->Remote Phone Book.
2. Select the desired value from the pull-down list of Search Remote Phonebook
Name.
3. Enter the desired time in the Search Flash Time (Seconds) field.
4. Click Confirm to accept the change.
LDAP (Lightweight Directory Access Protocol) is an application protocol for accessing and maintaining information services for the distributed directory over an IP network. IP phones can be configured to interface with a corporate directory server that supports
LDAP version 2 or 3 (Microsoft’s Active Directory is included).
The biggest plus for LDAP is that users can access the central LDAP directory of the corporation using IP phones, therefore they do not have to maintain the directory locally.
Users can search and dial out from the LDAP directory, and save LDAP entries to the local directory. LDAP entries displayed on the IP phone are read only, which cannot be added, edited or deleted by users. When an LDAP server is properly configured, the IP phone can look up entries from the LDAP server in a wide variety of ways. The LDAP server indexes all the data in its entries, and “filters” can be used to select the desired entry or group, and return the desired information.
Configurations on the IP phone limit the amount of the displayed entries when querying from the LDAP server, and decide how attributes are displayed and sorted.
Note LDAP feature is not applicable to SIP-T19P and SIP-T20P IP phones.
You can set a DSS key to be an LDAP key, and then press the LDAP key to enter the LDAP search screen when the IP phone is idle.
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LDAP Attributes
The following table lists the most common attributes used to configure the LDAP lookup on IP phones.
Abbreviation gn cn sn dn dc
-
- mobile ipPhone
Name givenName commonName surname distinguishedName dc company telephoneNumber mobilephoneNumber
IPphoneNumber
Description
First name
LDAP attribute is made up from given name joined to surname.
Last name or family name
Unique identifier for each entry
Domain component
Company or organization name
Office phone number
Mobile or cellular phone number
Home phone number
Procedure
LDAP can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure LDAP.
For more information, refer to
Assign an LDAP key.
For more information, refer to
Configure LDAP.
Navigate to: http://<phoneIPAddress>/servl et?p=contacts-LDAP&q=load
Assign an LDAP key.
Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&q=load&model=
0
Assign an LDAP key.
To configure LDAP via web user interface:
1. Click on Directory->LDAP.
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2. Enter the values in the corresponding fields.
3. Select the desired values from the corresponding pull-down lists.
4. Click Confirm to accept the change.
To configure an LDAP key via web user interface:
1. Click on DSSKey->Memory Key (Line Keys or Programable Key).
SIP-T19P IP phones only support programable keys and SIP-T22P/T21P/T20P IP phones only support line keys and programable keys.
2. In the desired DSS key field, select LDAP from the pull-down list of Type.
3. Click Confirm to accept the change.
To configure an LDAP key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
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3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select LDAP from the Key Type field.
5. Press the Save soft key to accept the change.
Busy Lamp Field (BLF) is used to monitor a specific user for status changes on IP phones.
For example, you can configure a BLF key on a supervisor’s phone to monitor the phone user status (busy or idle). When the monitored user places a call, a busy indicator on the supervisor’s phone indicates that the user’s phone is in use.
When the monitored user is idle, the supervisor presses the BLF key to dial out the phone number. When the monitored user receives an incoming call, the supervisor presses the
BLF key to pick up the call directly. When the monitored user is in a call, the supervisor presses the BLF key to interrupt and set up a conference call.
Note BLF feature is not applicable to SIP-T19P IP phones.
Visual Alert and Audio Alert for BLF Pickup
Visual and audio alert for BLF pickup allow the supervisor’s phone to play an alert tone and display a visual prompt (e.g., “6001<-6002”, 6001 is the monitored extension which receives an incoming call from 6002) when the monitored user receives an incoming call.
In addition to the BLF key, visual alert for BLF pickup feature enables the supervisor to pick up the monitored user’s incoming call by pressing the Pickup soft key. The directed call pickup code must be configured in advance. For more information on how to
configure the directed call pickup code for the Pickup soft key, refer to Directed Call
Note Visual alert for BLF pickup feature is not applicable to SIP-T20P IP phones.
BLF LED Mode
BLF LED Mode provides four kinds of definition for the BLF key LED status. The following table lists the LED statuses of the BLF key when BLF LED Mode is set to 0, 1, 2 or 3 respectively. The default value of BLF LED mode is 0.
Line key LED (configured as a BLF key and BLF LED Mode is set to 0)
LED Status
Solid green
Description
The monitored user is idle.
Fast flashing green (200ms) The monitored user receives an incoming call.
Slow flashing green (500ms)
The monitored user is dialing.
The monitored user is talking.
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Slow flashing green (1s)
Off
LED Status Description
The monitored user’s conversation is placed on hold.
The call is parked against the monitored user’s phone number.
The monitored user does not exist.
Memory key LED (configured as a BLF key and BLF LED Mode is set to 0)
LED Status
Solid green
Fast flashing red (200ms)
Solid red
Slow flashing red (1s)
Description
The monitored user is idle.
The monitored user receives an incoming call.
The monitored user is dialing.
The monitored user is talking.
The call is parked against the monitored user’s phone number.
The monitored user’s conversation is placed on hold.
The monitored user does not exist. Off
Line key LED (configured as a BLF key and BLF LED Mode is set to 1)
LED Status Description
Fast flashing green (200ms) The monitored user receives an incoming call.
Solid green
Slow flashing green (500ms)
Slow flashing green (1s)
Off
The monitored user is dialing.
The monitored user is talking.
The monitored user’s conversation is placed on hold.
The call is parked against the monitored user’s phone number.
The monitored user is idle.
The monitored user does not exist.
Memory key LED (configured as a BLF key and BLF LED Mode is set to 1)
LED Status
Fast flashing red (200ms)
Solid red
Slow flashing red (1s)
Description
The monitored user receives an incoming call.
The monitored user is dialing.
The monitored user is talking.
The call is parked against the monitored user’s phone number.
The monitored user’s conversation is placed on hold.
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LED Status
Off
Description
The monitored user is idle.
The monitored user does not exist.
Line key LED (configured as a BLF key and BLF LED Mode is set to 2)
LED Status Description
Fast flashing green (200ms) The monitored user receives an incoming call.
Slow flashing green (500ms)
Slow flashing green (1s)
Off
The monitored user is dialing.
The monitored user is talking.
The monitored user’s conversation is placed on hold.
The call is parked against the monitored user’s phone number.
The monitored user is idle.
The monitored user does not exist.
Memory key LED (configured as a BLF key and BLF LED Mode is set to 2)
LED Status
Fast flashing red (200ms)
Solid red
Slow flashing red (1s)
Off
Description
The monitored user receives an incoming call.
The monitored user is dialing.
The monitored user is talking.
The call is parked against the monitored user’s phone number.
The monitored user’s conversation is placed on hold.
The monitored user is idle.
The monitored user does not exist.
Line key LED (configured as a BLF key and BLF LED Mode is set to 3)
LED Status Description
Fast flashing green (200ms) The monitored user receives an incoming call.
Solid green
Slow flashing green (1s)
Off
The monitored user is dialing.
The monitored user is talking.
The monitored user’s conversation is placed on hold.
The call is parked against the monitored user’s phone number.
The monitored user is idle.
The monitored user does not exist.
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Memory key LED (configured as a BLF key and BLF LED Mode is set to 3)
LED Status
Fast flashing red (200ms)
Solid red
Slow flashing red (1s)
Off
Description
The monitored user receives an incoming call.
The monitored user is dialing.
The monitored user is talking.
The monitored user’s conversation is placed on hold.
The call is parked against the monitored user’s phone number.
The monitored user is idle.
The monitored user does not exist.
Procedure
BLF can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg y0000000000xx.cfg
Web User Interface
Specify whether to use visual alert and audio alert for BLF pickup.
For more information, refer to
Assign a BLF key.
For more information, refer to
on page
Configure BLF LED mode.
For more information, refer to
Assign a BLF key.
Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&q=load&model=
0
Specify whether to use visual alert and audio alert for BLF pickup.
Navigate to: http://<phoneIPAddress>/servl et?p=features-callpickup&q=lo ad
Configure BLF LED mode.
Navigate to:
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Phone User Interface http://<phoneIPAddress>/servl et?p=features-general&q=loa d
Assign a BLF key.
To configure a BLF key via web user interface:
1. Click on DSSKey->Memory Key (or Line Key).
2. In the desired DSS key field, select BLF from the pull-down list of Type.
3. Enter the phone number or extension you want to monitor in the Value field.
4. Select the desired line from the pull-down list of Line.
5. (Optional.) Enter the directed call pickup code in the Extension field.
6. Click Confirm to accept the change.
To configure visual alert and audio alert for BLF pickup via web user interface:
1. Click on Features->Call Pickup.
2. Select the desired value from the pull-down list of Visual Alert for BLF Pickup.
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Configuring Advanced Features
3. Select the desired value from the pull-down list of Audio Alert for BLF Pickup.
4. Click Confirm to accept the change.
To configure BLF LED mode via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of BLF LED Mode.
3. Click Confirm to accept the change.
To configure a BLF key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
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3. Press or , or the Switch soft key to select BLF from the Type field.
4. Press or , or the Switch soft key to select the desired line from the Account
ID field.
5. Enter the phone number or extension you want to monitor in the Value field.
6. (Optional.) Enter the directed call pickup code in the Extension field.
7. Press the Save soft key to accept the change.
Music on Hold (MoH) is the business practice of playing recorded music to fill the silence that would be heard by the party who has been placed on hold. To use this feature, specify a SIP URI pointing to an MoH server account. When a call is placed on hold, the IP phone will send an INVITE message to the specified MoH server account according to the SIP URI. The MoH server account automatically responds to the INVITE message and immediately plays audio from some source located anywhere (LAN,
Internet) to the held party.
Procedure
Music on hold can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure MoH on a per-line basis.
For more information, refer to
Configure MoH on a per-line basis.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
To configure MoH via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
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Configuring Advanced Features
4. Enter the SIP URI (e.g., sip:[email protected]) in the Music Server URI field.
5. Click Confirm to accept the change.
Automatic Call Distribution (ACD) enables organizations to manage a large number of phone calls on an individual basis. ACD enables the use of IP phones in a call-center role by automatically distributing incoming calls to available users, or agents. ACD depends on support from a SIP server. ACD is disabled on the phone by default. You need to enable it on a per-line basis before logging into the ACD system.
After the IP phone user logs into the ACD system, the server monitors the phone status and then decides whether to assign an incoming call to the user’s IP phone. When the phone status is changed to unavailable, the server stops distributing calls to the IP phone. The IP phone will remain in the unavailable status until the user manually changes the phone status or the ACD auto available timer (if configured) expires. How long the IP phone remains unavailable is configurable by auto-available timer. When the timer expires, the phone status is automatically changed to available. ACD auto available feature depends on support from a SIP server.
You need to configure an ACD key for the user to log into the ACD system. The ACD key
LED on the IP phone indicates the ACD status.
Note SIP-T19P IP phones support ACD feature for BroadSoft server only. For more information, refer to Yealink IP Phones Deployment Guide for BroadSoft UC-One Environments .
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Procedure
ACD can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Assign an ACD key.
For more information, refer to
Configure ACD auto available.
For more information, refer to
Assign an ACD key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&q=load&model=0
Configure ACD auto available.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-acd&q=load
Assign an ACD key.
To configure an ACD key via web user interface:
1. Click on DSSKey->Memory Key (or Line Key).
2. In the desired DSS key field, select ACD from the pull-down list of Type.
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3. Click Confirm to accept the change.
To configure ACD auto available via web user interface:
1. Click on Features->ACD.
2. Select the desired line from the pull-down list of ACD Auto Available.
Configuring Advanced Features
3. Enter the desired time in ACD Auto Available Timer (0~120s) field.
4. Click Confirm to accept the change.
To configure an ACD key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select ACD from the Type field.
4. Press the Save soft key to accept the change.
Message Waiting Indicator (MWI) informs users of the number of messages waiting in their mailbox without calling the mailbox. IP phones support both audio and visual MWI when receiving new voice messages.
IP phones support both solicited and unsolicited MWI. Unsolicited MWI is a server related feature.
The IP phone sends a SUBSCRIBE message to the server for message-summary updates.
The server sends a message-summary NOTIFY within the subscription dialog each time the MWI status changes. For solicited MWI, you must enable MWI subscription feature on IP phones. IP phones support subscribing the MWI messages to the account or the voice mail number.
IP phones do not need to subscribe for message-summary updates. The server automatically sends a message-summary NOTIFY in a new dialog each time the MWI status changes.
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Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure subscribe for MWI.
Configure subscribe MWI to voice mail.
For more information, refer to
Configure subscribe for MWI.
Configure subscribe MWI to voice mail.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
To configure subscribe for MWI via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
4. Select the desired value from the pull-down list of Subscribe for MWI.
5. Enter the period time in the MWI Subscription Period (Seconds) field.
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6. Click Confirm to accept the change.
To configure subscribe MWI to voice mail via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
4. Select the desired value from the pull-down list of Subscribe MWI To Voice Mail.
5. Enter the desired voice number in the Voice Mail field.
6. Click Confirm to accept the change.
Multicast paging allows IP phones to send/receive Real-time Transport Protocol (RTP) streams to/from the pre-configured multicast address(es) without involving SIP signaling.
Up to 10 listening multicast addresses can be specified on the IP phone.
Users can send an RTP stream without involving SIP signaling by pressing a configured multicast paging key. A multicast address (IP: Port) should be assigned to the multicast paging key, which is defined to transmit RTP stream to a group of designated IP phones.
When the IP phone sends the RTP stream to a pre-configured multicast address, each IP phone preconfigured to listen to the multicast address can receive the RTP stream.
When the originator stops sending the RTP stream, the subscribers stop receiving it. This feature is not applicable to SIP-T19P IP phones.
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Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Assign a multicast paging key.
For more information, refer to
Specify a multicast codec for the
IP phone to use for multicast RTP.
For more information, refer to
Sending RTP Stream on page 351 .
Web User Interface
Phone User Interface
Assign a multicast paging key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&q=load&model=0
Specify a multicast codec for the
IP phone to use for multicast RTP.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Assign a multicast paging key.
To configure a multicast paging key via web user interface:
1. Click on DSSKey->Memory Key (or Line Key).
2. In the desired DSS key field, select Multicast Paging from the pull-down list of Type.
3. Enter the multicast IP address and port number in the Value field.
The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
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4. Click Confirm to accept the change.
Configuring Advanced Features
To configure a codec for multicast paging via web user interface:
1. Click on Features->General Information.
2. Select the desired codec from the pull-down list of Multicast Codec.
3. Click Confirm to accept the change.
To configure a multicast paging key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Multicast Paging from the Key
Type field.
5. Enter the multicast IP address and port number in the Value field.
6. Press the Save soft key to accept the change.
IP phones can receive an RTP stream from the pre-configured multicast address(es) without involving SIP signaling, and can handle the incoming multicast paging calls differently depending on the configurations of Paging Barge and Paging Priority Active.
Paging Barge
This parameter defines the priority of the voice call in progress, and decides how the IP phone handles the incoming multicast paging calls when there is already a voice call in progress. If the parameter is configured as disabled, all incoming multicast paging calls
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Paging Priority Active
This parameter decides how the IP phone handles the incoming multicast paging calls when there is already a multicast paging call in progress. If the parameter is configured as disabled, the IP phone will automatically ignore all incoming multicast paging calls. If the parameter is configured as enabled, an incoming multicast paging call with higher priority is automatically answered, and the one with lower priority is ignored.
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Configure the listening multicast address.
Configure Paging Barge and
Paging Priority Active features.
For more information, refer to
on page
Web User Interface
Configure the listening multicast address.
Configure Paging Barge and
Paging Priority Active features.
Navigate to: http://<phoneIPAddress>/servlet
?p=contacts-multicastIP&q=load
To configure a listening multicast address via web user interface:
1. Click on Directory->Multicast IP.
2. Enter the listening multicast address and port number in the Listening Address field.
1 is the highest priority and 10 is the lowest priority.
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Configuring Advanced Features
3. Enter the label in the Label field.
The label will appear on the LCD screen when receiving the RTP multicast.
4. Click Confirm to accept the change.
To configure paging barge and paging priority active features via web user interface:
1. Click on Directory->Multicast IP.
2. Select the desired value from the pull-down list of Paging Barge.
3. Select the desired value from the pull-down list of Paging Priority Active.
4. Click Confirm to accept the change.
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Call recording enables users to record calls. It depends on support from a SIP server.
When the user presses the call record key, the IP phone sends a record request to the server. IP phones themselves do not have memory to store the recording, what they can do is to trigger the recording and indicate the recording status.
Normally, there are 2 main methods to trigger a recording on a certain server. We call them record and URL record. Record is for the IP phone to send the server a SIP INFO message containing a specific header. URL record is for the IP phone to send the server an HTTP GET message containing a specific URL. The server processes these messages and decides to start or stop a recording.
Note Call recording is not applicable to SIP-T19P IP phones.
Record
When a user presses a record key for the first time during a call, the IP phone sends a
SIP INFO message to the server with the specific header “Record: on”, and then the recording starts.
Example of a SIP INFO message:
Via: SIP/2.0/UDP 10.1.4.148:5063;branch=z9hG4bK1139980711
From: "827" <sip:[email protected]>;tag=2066430997
To:<sip:[email protected]>;tag=371745247
Call-ID: [email protected]
CSeq: 2 INFO
Contact: <sip:[email protected]:5063>
Max-Forwards: 70
User-Agent: Yealink SIP-T28P 2.72.0.1
Record: on
Content-Length: 0
When the user presses the record key for the second time, the IP phone sends a SIP
INFO message to the server with the specific header “Record: off”, and then the recording stops.
Example of a SIP INFO message:
Via: SIP/2.0/UDP 10.1.4.148:5063;branch=z9hG4bK1619489730
From: "827" <sip:[email protected]>;tag=1831694891
To:<sip:[email protected]>;tag=2228378244
Call-ID: [email protected]
CSeq: 3 INFO
Contact: <sip:[email protected]:5063>
Max-Forwards: 70
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User-Agent: Yealink SIP-T28P 2.72.0.1
Record: off
Content-Length: 0
URL Record
When a user presses a URL record key for the first time during a call, the IP phone sends an HTTP GET message to the server.
Example of an HTTP GET message:
Get /phonerecording.cgi?model=yealink HTTP/1.0\r\n
Request Method: GET
Request URI: /phonerecording.cgi?model=yealink
Request version: HTTP/1.0
Host: 10.1.2.224\r\n
User-agent: yealink SIP-T28P 2.72.0.1 00:16:65:11:30:68\r\n
If the recording is successfully started, the server will respond with a 200 OK message.
Example of a 200 OK message:
<YealinkIPPhoneText>
<Title>
</Title>
<Text>
The recording session is successfully started.
</Text>
<YealinkIPPhoneText>
If the recording fails for some reasons, for example, the recording box is full, the server will respond with a 200 OK message.
Example of a 200 OK message:
<YealinkIPPhoneText>
<Title>
</Title>
<Text>
Probably the recording box is full.
</Text>
<YealinkIPPhoneText>
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When the user presses the URL record key for the second time, the IP phone sends an
HTTP GET message to the server, and then the server will respond with a 200 OK message.
Example of a 200 OK message:
<YealinkIPPhoneText>
<Title>
</Title>
<Text>
The recording session is successfully stopped.
</Text>
<YealinkIPPhoneText>
Procedure
Call recording key can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Assign a record key.
For more information, refer to
Assign a URL record key.
For more information, refer to URL
Assign a record key and URL record key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&q=load&model=0
Assign a record key and URL record key.
To configure a record key via web user interface:
1. Click on DSSKey->Memory Key (or Line Key).
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2. In the desired DSS key field, select Record from the pull-down list of Type.
3. Click Confirm to accept the change.
To configure a URL record key via web user interface:
1. Click on DSSKey->Memory Key (or Line Key).
2. In the desired DSS key field, select URL Record from the pull-down list of Type.
3. Enter the URL in the Value field.
4. Click Confirm to accept the change.
To configure a record key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Record from the Key Type field.
5. Press the Save soft key to accept the change.
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To configure a URL record key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select URL Record from the Type field.
4. Enter the URL in the Value field.
5. Press the Save soft key to accept the change.
Hot desking originates from the definition of being the temporary physical occupant of a work station or surface by a particular employee. A primary motivation for hot desking is cost reduction. Hot desking is regularly used in places where not all employees are in the office at the same time, or not in the office for a long time, which means actual personal offices would often be vacant, consuming valuable space and resources.
Hot desking allows a user to clear registration configurations of all accounts on the IP phone, and then register his account on line 1. In order to use this feature, you need to assign a hot desking key.
Procedure
Hot desking key can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Assign a hot desking key.
For more information, refer to Hot
Assign a hot desking key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&q=load&model=0
Phone User Interface Assign a hot desking key.
To configure a hot desking key via web user interface:
1. Click on DSSKey->Memory Keys (Line Key or Programable Key).
SIP-T19P IP phones only support programable keys and SIP-T22P/T21P/T20P IP phones only support line keys and programable keys.
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2. In the desired DSS key field, select Hot Desking from the pull-down list of Type.
Note
3. Click Confirm to accept the change.
You can configure a programable key as a hot desking key on SIP-T19P IP phones only.
To configure a hot desking key via phone user interface:
1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Hot Desking from the Key Type field.
5. Press the Save soft key to accept the change.
Action URL allows IP phones to interact with web server applications by sending an
HTTP or HTTPS GET request. You can specify a URL that triggers a GET request when a specified event occurs. Action URL can only be triggered by the pre-defined events
(e.g., log on). The valid URL format is: http(s)://IP address of the server/help.xml?.
The following table lists the pre-defined events for action URL.
Event
Setup Completed
Registered
Unregistered
Register Failed
Off Hook
Description
When the IP phone completes startup.
When the IP phone successfully registers an account.
When the IP phone logs off the registered account.
When the IP phone fails to register an account.
When the IP phone is off hook.
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Event
On Hook
Incoming Call
Outgoing Call
Established
Terminated
Open DND
Close DND
Open Always Forward
Close Always Forward
Description
When the IP phone is on hook.
When the IP phone receives an incoming call.
When the IP phone places a call.
When the IP phone establishes a call.
When the IP phone terminates a call.
When the IP phone enables the DND mode.
When the IP phone disables the DND mode.
When the IP phone enables the always forward.
When the IP phone disables the always forward.
Open Busy Forward
Close Busy Forward
When the IP phone enables the busy forward.
When the IP phone disables the busy forward.
Open No Answer Forward When the IP phone enables the no answer forward.
Close No Answer Forward When the IP phone disables the no answer forward
Transfer Call
Blind Transfer
When the IP phone transfers a call.
When the IP phone blind transfers a call.
Attended Transfer
When the IP phone performs the semi-attended/attended transfer.
Hold
UnHold
Mute
UnMute
Missed Call
IP Changed
Forward Incoming Call
Reject Incoming Call
Answer New-In Call
Transfer Finished
Transfer Failed
Idle To Busy
Busy To Idle
When the IP phone places a call on hold.
When the IP phone retrieves a hold call.
When the IP phone mutes a call.
When the IP phone un-mutes a call.
When the IP phone misses a call.
When the IP address of the IP phone changes.
When the IP phone forwards an incoming call.
When the IP phone rejects an incoming call.
When the IP phone answers a new call.
When the IP phone completes to transfer a call.
When the IP phone fails to transfer a call.
When the state of the IP phone changes from idle to busy.
When the state of phone changes from busy to idle.
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An HTTP or HTTPS GET request may contain variable name and variable value, separated by “=”. Each variable value starts with $ in the query part of the URL. The valid URL format is: http(s)://IP address of server/help.xml?variable name=$variable.
Variable name can be customized by users, while the variable value is pre-defined. For example, a URL “
http://192.168.1.10/help.xml?mac=$mac
” is specified for the event
Mute, $mac will be dynamically replaced with the MAC address of the IP phone when the IP phone mutes a call.
The following table lists pre-defined variable values.
$mac
Variable Value
$ip
$model
$firmware
$active_url
$active_user
$active_host
$local
$remote
$display_local
$display_remote
Description
The MAC address of the IP phone
The IP address of the IP phone
The IP phone model
The firmware version of the IP phone
The SIP URI of the current account when the IP phone places a call, receives an incoming call or establishes a call.
The user part of the SIP URI for the current account when the IP phone places a call, receives an incoming call or establishes a call.
The host part of the SIP URI for the current account when the IP phone places a call, receives an incoming call or establishes a call.
The SIP URI of the caller when the IP phone places a call.
The SIP URI of the callee when the IP phone receives an incoming call.
The SIP URI of the callee when the IP phone places a call.
The SIP URI of the caller when the IP phone receives an incoming call.
The display name of the caller when the IP phone places a call.
The display name of the callee when the IP phone receives an incoming call.
The display name of the callee when the IP phone places a call.
The display name of the caller when the IP phone receives an incoming call.
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Variable Value
$call_id
Description
The call-id of the active call.
Procedure
Action URL can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure action URL.
For more information, refer to
Configure action URL.
Navigate to: http://<phoneIPAddress>/servl et?p=features-actionurl&q=loa d
To configure action URL via web user interface:
1. Click on Features->Action URL.
2. Enter the action URLs in the corresponding fields.
3. Click Confirm to accept the change.
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Opposite to action URL, action URI allows IP phones to interact with web server application by receiving and handling an HTTP or HTTPS GET request. When receiving a
GET request, the IP phone will perform the specified action and respond with a 200 OK message. A GET request may contain variable named as “key” and variable value,
Configuring Advanced Features which are separated by “=”. The valid URI format is: http(s)://phone IP address/servlet?key=variable value.
The following table lists pre-defined variable values:
Variable Value
OK
ENTER
SPEAKER
F_TRANSFER
VOLUME_UP
VOLUME_DOWN
MUTE
F_HOLD
X
0-9/*/POUND
L1-LX
D1-D10
F_CONFERENCE
F1-F4
MSG
HEADSET
RD
UP/DOWN/LEFT/RIGHT
Reboot
AutoP
DNDOn
DNDOff number=xxx&outgoing_uri=y
OFFHOOK
Phone Action
Press the OK key (For SIP-T19P, press ).
Press the Enter soft key (Except for SIP-T20P).
Press the Speakerphone key.
Transfers a call to another party.
Increase the volume.
Decrease the volume.
Mute the call.
Place an active call on hold.
Cancel actions or reject incoming calls (For
SIP-T22P/T21P/T20P, also mute or un-mute calls).
Press the keypad (0-9, * or #).
Press the line keys (Except for SIP-T19P, for
SIP-T28P, X=6, for SIP-T26/22P, X=3, for
SIP-T21P/T20P, X=2).
Press the memory keys (Only for SIP-T28/T26P).
Press the CONF key (Except for
SIP-T22P/T21P/T19P) or the Conference soft key
(Except for SIP-T20P).
Press the soft keys (Except for SIP-T20P).
Press the MESSAGE key.
Press the HEADSET key.
Press the RD key.
Press the navigation keys.
Reboot the IP phone.
Perform auto provisioning.
Activate the DND feature.
Deactivate the DND feature.
Place a call to xxx from SIP URI y.
Pick up the handset.
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ONHOOK
ANSWER
Reset
Variable Value
ATrans=xxx
BTrans=xxx
CALLEND
Phone Action
Hang up the handset.
Answer a call.
Reset a phone.
Perform a semi-attended/attended transfer to xxx.
Perform a blind transfer to xxx.
End a call. phonecfg=get[&accounts=x][&dnd
=x][&fw=x]
Get firmware version, registration, DND or forward configuration information.
The valid value of “x” is 0 or 1, 0 means you do not need to get configuration information. 1 means you want to get configuration information.
Note: The valid URI is: http(s)://phone IP address/servlet?phonecfg=get[&accounts=x][& dnd=x][&fw=x]
Note The variable value is not applicable to all events. For example, the variable value
“MUTE” is only applicable when the IP phone is during a call.
When authentication is required, you must enter
“p=login&q=login&username=xxx&pwd=yyy&jumpto=URI&” before the variable
“key”. xxx refers to the login user name and yyy refers to the login password.
For security reasons, IP phones do not receive and handle HTTP/HTTPS GET requests by default. You need to specify the trusted IP address for action URI. When the IP phone receives a GET request from the trusted IP address for the first time, the LCD screen prompts the message “Allow Remote Control?”. You can specify one or more trusted IP addresses on the IP phone, or configure the IP phone to receive and handle the URI from any IP address.
Procedure
Specify the trusted IP address for action URI using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Specify the trusted IP address(es) for sending the action URI to the IP phone.
For more information, refer to
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Configuring Advanced Features
Local Web User Interface
Specify the trusted IP address(es) for sending the action URI to the IP phone.
Navigate to: http://<phoneIPAddress>/servl et?p=features-remotecontrl&q
=load
To configure the trusted IP address(es) for action URI via web user interface:
1. Click on Features->Remote Control.
2. Enter the IP address or any in the Action URI allow IP List field.
Multiple IP addresses are separated by commas. If you enter “any” in this field, the
IP phone can receive and handle GET requests from any IP address. If you leave the field blank, the IP phone cannot receive or handle any HTTP GET request.
3. Click Confirm to accept the change.
Server redundancy is often required in VoIP deployments to ensure continuity of phone service, for events where the server needs to be taken offline for maintenance, the server fails, or the connection between the IP phone and the server fails.
Two types of redundancy are possible. In some cases, a combination of the two may be deployed:
Failover: In this mode, the full phone system functionality is preserved by having a second equivalent capability call server take over from the one that has gone down/off-line. This mode of operation should be done using the DNS mechanism from the primary to the secondary server.
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Fallback: In this mode, a second less featured call server with SIP capability takes over call control to provide basic calling capability, but without some advanced features offered by the working server (for example, shared line, call recording and MWI). IP phones support configuration of two SIP servers per SIP registration for fallback purpose.
Phone Configuration for Redundancy Implementation
To assist in explaining the redundancy behavior, an illustrative example of how an IP phone may be configured is shown as below. In the example, server redundancy for fallback and failover purposes is deployed. Two separate SIP servers (a working server and a fallback server) are configured for per line registration.
Working Server: Server 1 is configured with the domain name of the working server. For example, yealink.pbx.com. DNS mechanism is used such that the working server is resolved to multiple SIP servers for failover purpose. The working server is deployed in redundant pairs, designated as primary and secondary servers. The primary server has the highest priority server in a cluster of servers resolved by the DNS server. The secondary server backs up a primary server when the primary server fails and offers the same functionality as the primary server.
Fallback Server: Server 2 is configured with the IP address of the fallback server. For example, 192.168.1.15. A fallback server offers less functionality than the working server.
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Phone Registration
Registration methods of the fallback mode include:
Concurrent registration: The IP phone registers to two SIP servers (working server and fallback server) at the same time. In a failure situation, a fallback server can take over the basic calling capability, but without some of the advanced features offered by the working server (default registration method).
Successive registration: The IP phone only registers to one server at a time. The IP phone first registers to the working server. In a failure situation, the IP phone registers to the fallback server.
When registering to the working server, the IP phone must always register to the primary server first except in failover conditions. When the primary server registration is unavailable, the secondary server will serve as the working server.
Procedure
Server redundancy can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure the server redundancy on the IP phone.
For more information, refer to
on page
Configure the server redundancy on the IP phone.
Navigate to: http://<phoneIPAddress>/servl et?p=account-register&q=load
&acc=0
To configure server redundancy for fallback purpose via web user interface:
1. Click on Account->Register.
2. Select the desired account from the pull-down list of Account.
3. Configure registration parameters of the selected account in the corresponding fields.
4. Select the desired value from the pull-down list of Transport.
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5. Configure parameters of SIP server 1 and SIP server 2 in the corresponding fields.
6. Click Confirm to accept the change.
To configure server redundancy for failover purpose via web user interface:
1. Click on Account->Register.
2. Select the desired account from the pull-down list of Account.
3. Configure registration parameters of the selected account in the corresponding fields.
4. Select DNS-NAPTR from the pull-down list of Transport.
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5. Configure parameters of the SIP server 1 or SIP server 2 in the corresponding fields.
You must set the port of SIP server to 0 for NAPTR, SRV and A queries.
Note
6. Click Confirm to accept the change.
If the outbound proxy server is required and the transport is set to DNS-NAPTR, you must set the port of outbound proxy server to 0 for NAPTR, SRV and A queries.
If a domain name is configured for a SIP server, the IP address(es) associated with that domain name will be resolved through DNS as specified by RFC 3263. The DNS query involves NAPTR, SRV and A queries, which allows the IP phone to adapt to various deployment environments. The IP phone performs NAPTR query for the NAPTR pointer and transport protocol (UDP, TCP and TLS), the SRV query on the record returned from the NAPTR for the target domain name and the port number, and the A query for the IP addresses.
If an explicit port (except 0) is specified and the transport type is set to DNS-NAPTR, A query will be performed only. If a SIP server port is set to 0 and the transport type is set to DNS-NAPTR, NAPTR and SRV queries will be tried before falling to A query. If no port is found through the DNS query, 5060 will be used.
The following details the procedures of DNS query for the IP phone to resolve the domain name (e.g., yealink.pbx.com) of working server into the IP address, port and transport protocol.
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NAPTR (Naming Authority Pointer)
First, the IP phone sends NAPTR query to get the NAPTR pointer and transport protocol.
Example of NAPTR records: order pref flags service regexp replacement
IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.yealink.pbx.com
IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.yealink.pbx.com
Parameters are explained in the following table:
Parameter order pref
Description
Specify preferential treatment for the specific record. The order is from lowest to highest, lower order is more preferred.
Specify the preference for processing multiple NAPTR records with the same order value. Lower value is more preferred. flags The flag “s” means to perform an SRV lookup. service
Specify the transport protocols:
SIP+D2U: SIP over UDP
SIP+D2T: SIP over TCP
SIP+D2S: SIP over SCTP
SIPS+D2T: SIPS over TCP regexp Always empty for SIP services. replacement Specify a domain name for the next query.
The IP phone picks the first record, because its order of 90 is lower than 100. The pref parameter is unimportant as there is no other record with order 90. The flag “s” indicates performing the SRV query next. TCP will be used, targeted to a host determined by an SRV query of “_sip._tcp.yealink.pbx.com”. If the flag of the NAPTR record returned is empty, the IP phone will perform NAPTR query again according to the previous NAPTR query result.
SRV (Service Location Record)
The IP phone performs an SRV query on the record returned from the NAPTR for the host name and the port number. Example of SRV records:
Priority Weight Port Target
IN SRV 0 1 5060 server1.yealink.pbx.com
IN SRV 0 2 5060 server2.yealink.pbx.com
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Parameters are explained in the following table:
Parameter
Priority
Weight
Description
Specify preferential treatment for the specific host entry. Lower priority is more preferred.
When priorities are equal, weight is used to differentiate the preference. The preference is from highest to lowest. Keep the same to load balance.
Identify the port number to be used.
Identify the actual host for an A query.
Port
Target
SRV query returns two records. The two SRV records point to different hosts and have the same priority 0. The weight of the second record is higher than the first one, so the second record will be picked first. The two records also contain a port “5060”, the IP phone uses this port. If the Target is not a numeric IP address, the IP phone performs an
A query. So in this case, the IP phone uses “server1.yealink.pbx.com" and
“server2.yealink.pbx.com" for the A query.
A (Host IP Address)
The IP phone performs an A query for the IP address of each target host name. Example of A records:
Server1.yealink.pbx.com IN A 192.168.1.13
Server2.yealink.pbx.com IN A 192.168.1.14
The IP phone picks the IP address “192.168.1.14” first.
Outgoing Call When the Working Server Connection Fails
When a user initiates a call, the phone will go through the following steps to connect the call:
1. Sends the INVITE request to the primary server.
2. If the primary server does not respond correctly to the INVITE, then tries to make the call using the secondary server.
3. If the secondary server is also unavailable, the IP phone will try the fallback server until it either succeeds in making a call or exhausts all servers at which point the call will fail.
At the start of a call, server availability is determined by SIP signaling failure. SIP signaling failure depends on the SIP protocol being used as described below:
If TCP is used, then the signaling fails if the connection or the send fails.
If UDP is used, then the signaling fails if ICMP is detected or if the signal times out. If the signaling has been attempted through all servers in the list and this is the last server, then the signaling fails after the complete UDP timeout defined in RFC 3261.
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If it is not the last server in the list, the maximum number of retries depends on the configured retry count.
Procedure
Server redundancy can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure the transport type on the IP phone.
For more information, refer to
Configure the transport type on the IP phone.
Navigate to: http://<phoneIPAddress>/servl et?p=account-register&q=load
&acc=0
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LLDP (Linker Layer Discovery Protocol) is a vendor-neutral Link Layer protocol, which allows IP phones to receive and/or transmit device-related information from/to directly connected devices on the network that are also using the protocol, and store the information about other devices. LLDP transmits information as packets called LLDP
Data Units (LLDPDUs). An LLDPDU consists of a set of Type-Length-Value (TLV) elements, each of which contains a particular type of information about the device or the port transmitting it.
LLDP-MED (Media Endpoint Discovery)
LLDP-MED is published by the Telecommunications Industry Association (TIA). It is an extension to LLDP that operates between endpoint devices and network connectivity devices. LLDP-MED specifically provides support for voice over IP (VoIP) applications and provides the following capabilities:
Capabilities Discovery -- allows IP phones to determine the capabilities that the connected switch supports and has enabled.
Network Policy -- provides voice VLAN configuration to notify IP phones which VLAN to use and QoS-related configuration for voice data. It provides a “plug and play” network environment.
Power Management -- provides information related to how IP phones are powered, power priority, and how much power IP phones need.
Inventory Management -- provides a means to effectively manage IP phones and
Configuring Advanced Features their attributes such as model number, serial number and software revision.
TLVs supported by IP phones are summarized in the following table:
TLV Type TLV Name
Chassis ID
Mandatory TLVs
Port ID
Time To Live
End of LLDPDU
Description
The network address of the IP phone.
The MAC address of the IP phone.
Seconds until data unit expires.
Marks end of LLDPDU.
Optional TLVs
System Name
System Description
Name assigned to the IP phone.
The default value is “yealink”.
Description of the IP phone.
The default value is “yealink”.
System Capabilities
The supported and enabled capabilities of the IP phone.
The supported capabilities are Bridge,
Telephone and Router.
The enabled capabilities are Bridge and
Telephone by default.
Port Description
Description of port that sends data unit.
The default value is “WAN PORT”.
IEEE Std 802.3
Organizationally
Specific TLV
MAC/PHY
Configuration/Status
Duplex and bit rate settings of the IP phone.
The Auto Negotiation is supported and enabled by default.
The advertised capabilities of PMD.
Auto-Negotiation is: 100BASE-TX (full duplex mode), 100BASE-TX (half duplex mode), 10BASE-T (full duplex mode), or
10BASE-T (half duplex mode).
TIA
Organizationally
Specific TLVs
Media Capabilities
The MED device type of the IP phone and the supported LLDP-MED TLV type can be encapsulated in LLDPDU.
The supported LLDP-MED TLV types are:
LLDP-MED Capabilities, Network Policy,
Extended Power via MDI-PD and
Inventory.
Network Policy Port VLAN ID, application type, L2 priority
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TLV Type TLV Name Description and DSCP value.
Extended
Power-via-MDI
Inventory – Model
Name
Power type, source, priority and value.
Inventory –
Hardware Revision
Inventory –
Firmware Revision
Inventory –
Software Revision
Inventory – Serial
Number
Hardware revision of the IP phone.
Firmware revision of the IP phone.
Software revision of the IP phone.
Serial number of the IP phone.
Inventory –
Manufacturer Name
Manufacturer name of the IP phone.
The default value is “yealink”.
Model name of the IP phone.
Asset ID
Assertion identifier of the IP phone.
The default value is “asset”.
Procedure
LLDP can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure LLDP.
For more information, refer to
Configure LLDP.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
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Configuring Advanced Features
To configure LLDP via web user interface:
1. Click on Network->Advanced.
2. In the LLDP block, select the desired value from the pull-down list of Active.
3. Enter the desired time interval in the Packet Interval (1~3600s) field.
4. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
5. Click OK to reboot the IP phone.
VLAN (Virtual Local Area Network) is used to logically divide a physical network into several broadcast domains. VLAN membership can be configured through software instead of physically relocating devices or connections. Grouping devices with a common set of requirements regardless of their physical location can greatly simplify network design. VLANs can address issues such as scalability, security and network management.
The purpose of VLAN configurations on the IP phone is to insert tag with VLAN information to the packets generated by the IP phone. When VLAN is properly configured for the ports (Internet port and PC port) on the IP phone, the IP phone will tag all packets from these ports with the VLAN ID. The switch receives and forwards the tagged packets to the corresponding VLAN according to the VLAN ID in the tag as described in IEEE Std 802.3.
VLAN on IP phones allows simultaneous access for a regular PC. This feature allows a PC to be daisy chained to an IP phone and the connection for both PC and IP phone to be trunked through the same physical Ethernet cable.
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In addition to manual configuration, the IP phone also supports automatic discovery of
VLAN via LLDP or DHCP. The assignment takes place in this order: assignment via LLDP, manual configuration, then assignment via DHCP.
VLAN Discovery via DHCP
IP phones support VLAN discovery via DHCP. When the VLAN Discovery method is set to
DHCP, the IP phone will examine DHCP option for a valid VLAN ID. The predefined option 132 is used to supply the VLAN ID by default. You can customize the DHCP option used to request the VLAN ID.
Procedure
VLAN can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure VLAN for the Internet port and PC port manually.
Configure DHCP VLAN discovery feature.
For more information, refer to
Configure VLAN for the Internet port and PC port.
Configure DHCP VLAN discovery feature.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
Configure VLAN for the Internet port and PC port.
To configure VLAN for Internet port via web user interface:
1. Click on Network->Advanced.
2. In the VLAN block, select the desired value from the pull-down list of WAN Port
Active.
3. Enter the VLAN ID in the VID (1-4094) field.
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Configuring Advanced Features
4. Select the desired value (0-7) from the pull-down list of Priority.
5. Click Confirm to accept the change.
A dialog box pops up to prompt that the settings will take effect after a reboot.
6. Click OK to reboot the IP phone.
To configure VLAN for PC port via web user interface:
1. Click on Network->Advanced.
2. In the VLAN block, select the desired value from the pull-down list of PC Port Active.
3. Enter the VLAN ID in the VID (1-4094) field.
4. Select the desired value (0-7) from the pull-down list of Priority.
5. Click Confirm to accept the change.
A dialog box pops up to prompt that the settings will take effect after a reboot.
6. Click OK to reboot the IP phone.
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To configure DHCP VLAN discovery via web user interface:
1. Click on Network->Advanced.
2. In the VLAN block, select the desired value from the pull-down list of DHCP VLAN
Active.
3. Enter the desired option in the Option field.
The default option is 132.
4. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
5. Click OK to reboot the IP phone.
To configure VLAN for Internet port (or PC port) via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin)
->Network->VLAN->WAN Port (or PC Port).
2. Press or , or the Switch soft key to select the desired value from the VLAN
Status field.
3. Enter the VLAN ID (1-4094) in the VID Number field.
4. Enter the priority value (0-7) in the Priority field.
5. Press the Save soft key to accept the change
The IP phone reboots automatically to make settings effective after a period of time.
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VPN (Virtual Private Network) is a secured private network connection built on top of public telecommunication infrastructure, such as the Internet. It has become more prevalent due to benefits of scalability, reliability, convenience and security. VPN
Configuring Advanced Features provides remote offices or individual users with secure access to their organization's network. There are two types of VPN access: remote-access VPN (connecting an individual device to a network) and site-to-site VPN (connecting two networks together).
Remote-access VPN allows employees to access their company's intranet from home or outside the office, and site-to-site VPN allows employees in geographically separated offices to share one cohesive virtual network. VPN can be also classified by the protocols used to tunnel the traffic. It provides security through tunneling protocols:
IPSec, SSL, L2TP and PPTP.
IP phones support SSL VPN, which provides remote-access VPN capabilities through SSL.
OpenVPN is a full featured SSL VPN software solution that creates secure connections in remote access facilities, designed to work with the TUN/TAP virtual network interface.
TUN and TAP are virtual network kernel devices. TAP simulates a link layer device and provides a virtual point-to-point connection, while TUN simulates a network layer device and provides a virtual network segment. IP phones use OpenVPN to achieve VPN feature. To prevent disclosure of private information, tunnel endpoints must authenticate each other before secure VPN tunnel is established. After VPN feature is configured properly on the IP phone, the IP phone acts as a VPN client and uses the certificates to authenticate the VPN server.
To use VPN, the compressed package of VPN-related files should be uploaded to the IP phone in advance. The file format of the compressed package must be *.tar. The related VPN files are: certificates (ca.crt and client.crt), key (client.key) and the configuration file (vpn.cnf) of the VPN client. For more information on how to package a
TAR file, refer to
OpenVPN Feature on Yealink IP Phones
.
Note VPN feature is not applicable to SIP-T19P IP phones.
Procedure
VPN can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure VPN feature and upload a TAR file to the IP phone.
For more information, refer to
Configure VPN feature and upload a TAR package to the IP phone.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
Phone User Interface Configure VPN feature.
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To upload a TAR file and configure VPN via web user interface:
1. Click on Network->Advanced.
2. Click Browse to locate the TAR file from the local system.
3. Click Upload to upload the TAR file.
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The web user interface prompts the message “Import config…”.
4. In the VPN block, select the desired value from the pull-down list of Active.
5. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
6. Click OK to reboot the IP phone.
To configure VPN via phone user interface after uploading a TAR file:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->VPN.
2. Press or , or the Switch soft key to select the desired value from the VPN
Active field.
You must upload the OpenVPN TAR file using configuration files or via web user interface in advance.
3. Press the Save soft key to accept the change.
The IP phone reboots automatically to make settings effective after a period of time.
Configuring Advanced Features
Quality of Service (QoS) is the ability to provide different priorities for different packets in the network, allowing the transport of traffic with special requirements. QoS guarantees are important for applications that require fixed bit rate and are delay sensitive when the network capacity is insufficient. There are four major QoS factors to be considered when configuring a modern QoS implementation: bandwidth, delay, jitter and loss.
QoS provides better network service through the following features:
Supporting dedicated bandwidth
Improving loss characteristics
Avoiding and managing network congestion
Shaping network traffic
Setting traffic priorities across the network
The Best-Effort service is the default QoS model in IP networks. It provides no guarantees for data delivering, which means delay, jitter, packet loss and bandwidth allocation are unpredictable. Differentiated Services (DiffServ or DS) is the most widely used QoS model. It provides a simple and scalable mechanism for classifying and managing network traffic and providing QoS on modern IP networks. Differentiated
Services Code Point (DSCP) is used to define DiffServ classes and stored in the first six bits of the ToS (Type of Service) field. Each router on the network can provide QoS simply based on the DiffServ class. The DSCP value ranges from 0 to 63 with each DSCP specifying a particular per-hop behavior (PHB) applicable to a packet. A PHB refers to the packet scheduling, queuing, policing, or shaping behavior of a node on any given packet.
Four standard PHBs available to construct a DiffServ-enabled network and achieve
QoS:
Class Selector PHB -- backwards compatible with IP precedence. Class Selector code points are of the form “xxx000”. The first three bits are the IP precedence bits.
These class selector PHBs retain almost the same forwarding behavior as nodes that implement IP precedence-based classification and forwarding.
Expedited Forwarding PHB -- the key ingredient in DiffServ model for providing a low-loss, low-latency, low-jitter and assured bandwidth service.
Assured Forwarding PHB -- defines a method by which BAs (Bandwidth Allocations) can be given different forwarding assurances.
Default PHB -- specifies that a packet marked with a DSCP value of “000000” gets the traditional best effort service from a DS-compliant node.
VoIP is extremely bandwidth- and delay-sensitive. QoS is a major issue in VoIP implementations, regarding how to guarantee that packet traffic not be delayed or
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Voice QoS
In order to make VoIP transmissions intelligible to receivers, voice packets should not be dropped, excessively delayed, or made to suffer varying delay. DiffServ model can guarantee high-quality voice transmission when the voice packets are configured to a higher DSCP value.
SIP QoS
SIP protocol is used for creating, modifying and terminating two-party or multi-party sessions. To ensure good voice quality, SIP packets emanated from IP phones should be configured with a high transmission priority.
DSCPs for voice and SIP packets can be specified respectively.
Procedure
QoS can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the DSCPs for voice packets and SIP packets.
For more information, refer to
Configure the DSCPs for voice packets and SIP packets.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
To configure DSCPs for voice packets and SIP packets via web user interface:
1. Click on Network->Advanced.
2. Enter the desired value in the Voice QoS (0~63) field.
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3. Enter the desired value in the SIP QoS (0~63) field.
4. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
5. Click OK to reboot the IP phone.
Network Address Translation (NAT) is essentially a translation table that maps public IP address and port combinations to private ones. This reduces the need for a large number of public IP addresses. NAT ensures security since each outgoing or incoming request must first go through a translation process. But in the VoIP environment, NAT breaks end-to-end connectivity.
NAT Traversal
NAT traversal is a general term for techniques that establish and maintain IP connections traversing NAT gateways, typically required for client-to-client networking applications, especially for VoIP deployments. STUN is one of the NAT traversal techniques supported by IP phones.
STUN (Simple Traversal of UDP over NATs)
STUN is a network protocol, used in NAT traversal for applications of real-time voice, video, messaging, and other interactive IP communications. The STUN protocol allows applications to operate behind a NAT to discover the presence of the network address translator, and to obtain the mapped (public) IP address and port number that the NAT has allocated for the UDP connections to remote parties. The protocol requires
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Internet. The IP phone can be configured to act as a STUN client, to send exploratory
STUN messages to the STUN server. The STUN server uses those messages to determine the public IP address and port used, and then informs the client.
The NAT traversal and STUN server are configurable on a per-line basis.
Procedure
NAT traversal and STUN server can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure NAT traversal and
STUN server on the IP phone.
For more information, refer to
on page
Configure NAT traversal and
STUN server on the IP phone.
Navigate to: http://<phoneIPAddress>/servl et?p=account-register&q=load
&acc=0
To configure NAT traversal and STUN server via web user interface:
1. Click on Account->Register.
2. Select the desired account from the pull-down list of Account.
3. Select STUN from the pull-down list of NAT.
4. Enter the IP address or the domain name of the STUN server in the STUN Server field.
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5. Click Confirm to accept the change.
Configuring Advanced Features
IEEE 802.1X authentication is an IEEE standard for Port-based Network Access Control
(PNAC), part of the IEEE 802.1 group of networking protocols. It offers an authentication mechanism for devices to connect/link to a LAN or WLAN. The 802.1X authentication involves three parties: a supplicant, an authenticator and an authentication server. The supplicant is the IP phone that wishes to attach to the LAN or WLAN. With 802.1X port-based authentication, the IP phone provides credentials, such as user name and password, for the authenticator, and then the authenticator forwards the credentials to the authentication server for verification. If the authentication server determines the credentials are valid, the IP phone is allowed to access resources located on the protected side of the network.
IP phones support protocols EAP-MD5, EAP-TLS, PEAP-MSCHAPv2 and
EAP-TTLS/EAP-MSCHAPv2 for 802.1X authentication.
Procedure
802.1X authentication can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure the 802.1X authentication.
For more information, refer to
on page
Configure the 802.1X authentication.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
Configure the 802.1X authentication.
To configure the 802.1X authentication via web user interface:
1. Click on Network->Advanced.
2. In the 802.1x block, select the desired protocol from the pull-down list of 802.1x
Mode. a) If you select EAP-MD5:
1) Enter the user name for authentication in the Identity field.
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2) Enter the password for authentication in the MD5 Password field. b) If you select EAP-TLS:
1) Enter the user name for authentication in the Identity field.
2) Leave the MD5 Password field blank.
3) In the CA Certificates field, click Browse to select the desired CA certificate
(*.pem, *.crt, *.cer or *.der) from your local system.
4) In the Device Certificates field, click Browse to select the desired client (*.pem or *.cer) certificate from your local system.
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Configuring Advanced Features
5) Click Upload to upload the certificates. c) If you select PEAP-MSCHAPv2:
1) Enter the user name for authentication in the Identity field.
2) Enter the password for authentication in the MD5 Password field.
3) In the CA Certificates field, click Browse to select the desired CA certificate
(*.pem, *.crt, *.cer or *.der) from your local system.
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4) Click Upload to upload the certificate. d) If you select EAP-TTLS/EAP-MSCHAPv2:
1) Enter the user name for authentication in the Identity field.
2) Enter the password for authentication in the MD5 Password field.
3) In the CA Certificates field, click Browse to select the desired CA certificate
(*.pem, *.crt, *.cer or *.der) from your local system.
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4) Click Upload to upload the certificate.
3. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after a reboot.
4. Click OK to reboot the IP phone.
To configure the 802.1X authentication via phone user interface after:
1. Press Menu->Settings->Advanced Settings (password: admin)
->Network->802.1x Settings.
2. Press or , or the Switch soft key to select the desired value from the 802.1x
Mode field. a) If you select EAP-MD5:
1) Enter the user name for authentication in the Identity field.
2) Enter the password for authentication in the MD5 Password field. b) If you select EAP-TLS:
1) Enter the user name for authentication in the Identity field.
2) Leave the MD5 Password field blank. c) If you select PEAP-MSCHAPv2:
1) Enter the user name for authentication in the Identity field.
2) Enter the password for authentication in the MD5 Password field. d) If you select EAP-TTLS/EAP-MSCHAPv2:
1) Enter the user name for authentication in the Identity field.
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2) Enter the password for authentication in the MD5 Password field.
3. Click Save to accept the change.
The IP phone reboots automatically to make the settings effective after a period of time.
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TR-069 is a technical specification defined by the Broadband Forum, which defines a mechanism that encompasses secure auto-configuration of a CPE (Customer-Premises
Equipment), and incorporates other CPE management functions into a common framework. TR-069 uses common transport mechanisms (HTTP and HTTPS) for communication between CPE and ACS (Auto Configuration Servers). The HTTP(S) messages contain XML-RPC methods defined in the standard for configuration and management of the CPE.
TR-069 is intended to support a variety of functionalities to manage a collection of CPEs, including the following primary capabilities:
Auto-configuration and dynamic service provisioning
Software or firmware image management
Status and performance monitoring
Diagnostics
The following table provides a description of RPC methods supported by IP phones.
RPC Method
GetRPCMethods
SetParameterValues
GetParameterValues
GetParameterNames
GetParameterAttributes
SetParameterAttributes
Reboot
Download
Description
This method is used to discover the set of methods supported by the CPE.
This method is used to modify the value of one or more CPE parameters.
This method is used to obtain the value of one or more CPE parameters.
This method is used to discover the parameters accessible on a particular CPE.
This method is used to read the attributes associated with one or more CPE parameters.
This method is used to modify attributes associated with one or more CPE parameters.
This method causes the CPE to reboot.
This method is used to cause the CPE to download a specified file from the designated location.
Configuring Advanced Features
RPC Method
Upload
ScheduleInform
FactoryReset
TransferComplete
AddObject
DeleteObject
Description
File types supported by IP phones are:
Firmware Image
Configuration File
This method is used to cause the CPE to upload a specified file to the designated location.
File types supported by IP phones are:
Configuration File
Log File
This method is used to request the CPE to schedule a one-time Inform method call (separate from its periodic Inform method calls) sometime in the future.
This method resets the CPE to its factory default state.
This method informs the ACS of the completion
(either successful or unsuccessful) of a file transfer initiated by an earlier Download or Upload method call.
This method is used to add a new instance of an object defined on the CPE.
This method is used to remove a particular instance of an object.
Procedure
TR-069 can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure TR-069 feature.
For more information, refer to
on page
Configure TR-069 feature.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-preference&q=lo ad
To configure TR-069 via web user interface:
1. Click on Settings->TR069.
2. Select Enabled from the pull-down list of Enable TR069.
3. Enter the user name and password authenticated by the ACS in the ACS Username and ACS Password fields.
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4. Enter the URL of the ACS in the ACS URL field.
5. Select the desired value from the pull-down list of Enable Periodic Inform.
6. Enter the desired time in the Periodic Inform Interval (seconds) field.
7. Enter the user name and password authenticated by the IP phone in the
Connection Request Username and Connection Request Password fields.
8. Click Confirm to accept the change.
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IPv6 is the next generation network layer protocol, designed as a replacement for the current IPv4 protocol. IPv6 is developed by the Internet Engineering Task Force (IETF) to deal with the long-anticipated problem of IPv4 address exhaustion. IPv6 uses a 128-bit address, consisting of eight groups of four hexadecimal digits separated by colons. VoIP network based on IPv6 can ensure QoS, a set of service requirements to deliver performance guarantee while transporting traffic over the network.
IPv6 Address Assignment Method
Supported IPv6 address assignment methods:
Manual Assignment: An IPv6 address and other configuration parameters (e.g.,
DNS server) for the IP phone can be statically configured by an administrator.
Stateless Address Autoconfiguration (SLAAC): SLAAC is one of the most convenient methods to assign IP addresses to IPv6 nodes. SLAAC requires no manual configuration of the IP phone, minimal (if any) configuration of routers, and no additional servers. To use IPv6 SLAAC, the IP phone must be connected to a network with at least one IPv6 router connected. This router is configured by the network administrator and sends out Router Advertisement announcements onto the link. These announcements can allow the on-link connected IP phone to configure itself with IPv6 address, as specified in RFC 4862.
Configuring Advanced Features
Procedure
IPv6 can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the IPv6 address assignment method.
For more information, refer to
Configure the IPv6 address assignment method.
Navigate to: http://<phoneIPAddress>/servl et?p=network&q=load
To configure IPv6 address assignment method via web user interface:
1. Click on Network->Basic.
2. Select the desired address mode (IPv6 or IPv4&IPv6) from the pull-down list of
Mode (IPv4/IPv6).
3. In the IPv6 Config block, do one of the following.
- If you mark the Static IP Address radio box, configure the IPv6 address and other configuration parameters in the corresponding fields.
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- (Optional.) If you mark the DHCP radio box, you can configure the static DNS address in the corresponding fields.
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4. Click Confirm to accept the change.
A dialog box pops up to prompt that the settings will take effect after a reboot.
5. Click OK to reboot the IP phone.
To configure IPv6 address assignment method via phone user interface:
1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN
Port.
2. Press or to select IPv4&IPv6 or IPv6 from the IP Mode field.
3. Press or to highlight IPv6 and press the Enter soft key.
4. Press or to select the desired IPv6 address assignment method.
If you select the Static IPv6 Client, configure the IPv6 address and other network parameters in the corresponding fields.
5. Press the Save soft key to accept the change
The IP phone reboots automatically to make settings effective after a period of time.
Configuring Audio Features
This chapter provides information for making configuration changes for the following audio features:
Headset prior allows users to use headset preferentially if a headset is physically connected to the IP phone. This feature is especially useful for permanent or full-time headset users.
Procedure
Headset prior can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure headset prior.
For more information, refer to
Configure headset prior.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
To configure headset prior via web user interface:
1. Click on Features->General Information.
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2. Select the desired value from the pull-down list of Headset Prior.
3. Click Confirm to accept the change.
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Dual headset allows users to use two headsets on one IP phone. To use this feature, users need to physically connect two headsets to the headset and handset jacks respectively. Once the phone connects to a call, the user with the headset connected to the headset jack has full-duplex capabilities, while the user with the headset connected to the handset jack is only able to listen. This feature is not applicable to SIP-T19P and
SIP-T21P IP phones.
Procedure
Dual headset can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure dual headset.
For more information, refer to
Configure dual headset.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
To configure dual headset via web user interface:
1. Click on Features->General Information.
Configuring Audio Features
2. Select the desired value from the pull-down list of Dual-Headset.
3. Click Confirm to accept the change.
CODEC is an abbreviation of COmpress-DECompress, capable of coding or decoding a digital data stream or signal by implementing an algorithm. The object of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining the quality. This can effectively reduce the frame size and the bandwidth required for audio transmission.
The default codecs used on IP phones are summarized in the following table:
Codec
PCMA
PCMU
G729
G722
Algorithm
G.711 a-law
G.711 u-law
G.729
G.722
Bit Rate
64 Kbps
64 Kbps
8 Kbps
64 Kbps
Sample Rate Packetization Time
8 Ksps
8 Ksps
20ms
20ms
8 Ksps 20ms
16 Ksps 20ms
In addition to the codecs introduced above, IP phones also support codecs: G723_53,
G723_63, G726-16, G726-24, G726-32, G726-40 (Codecs G726-16, G726-24 and G726-40 are not applicable to SIP-T21P and SIP-T19P IP phones). Codecs and priorities of these codecs are configurable on a per-line basis. The attribute “rtpmap” is used to define a mapping from RTP payload codes to a codec, clock rate and other encoding parameters.
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The corresponding attributes of the codec are listed as follows:
Codec Priority
PCMU
PCMA
G729
G722
G723_53
G723_63
G726-16
G726-24
G726-32
G726-40 iLBC
Configuration Methods
Configuration Files
Web User Interface
Configuration Files
Web User Interface
Configuration Files
Web User Interface
Configuration Files
Web User Interface
Configuration Files
Web User Interface
Configuration Files
Web User Interface
Configuration Files
Web User Interface
Configuration Files
Web User Interface
Configuration Files
Web User Interface
Configuration Files
Web User Interface
Configuration Files
Web User Interface
1
2
3
4
0
0
0
0
0
0
0
RTPmap
0
8
18
9
4
4
103
104
102
105
106
Packetization Time
Ptime (Packetization Time) is a measurement of the duration (in milliseconds) of the audio data in each RTP packet sent to the destination, and defines how much network bandwidth is used for the RTP stream transfer. Before establishing a conversation, codec and ptime are negotiated through SIP signaling. The valid values of ptime range from
10 to 60, in increments of 10 milliseconds. The default ptime is 20ms. You can also disable the ptime negotiation.
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Configuring Audio Features
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure the codecs to use on a per-line basis.
Configure the priority and rtpmap for the enabled codec.
For more information, refer to
Configure the ptime.
For more information, refer to
Configure the codecs to use and adjust the priority of the enabled codecs on a per-line basis.
Configure the ptime.
Navigate to: http://<phoneIPAddress>/servl et?p=account-codec&q=load& acc=0
To configure the codecs to use and adjust the priority of the enabled codecs on a per-line basis via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Codec.
4. Select the desired codec from the Disable Codecs column and then click .
The selected codec appears in the Enable Codecs column.
5. Repeat the step 4 to add more codecs to the Enable Codecs column.
6. To remove the codec from the Enable Codecs column, select the desired codec and then click .
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7. To adjust the priority of codecs, select the desired codec and then click or .
8. Click Confirm to accept the change.
To configure the ptime on a per-line basis via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
4. Select the desired value from the pull-down list of PTime (ms).
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5. Click Confirm to accept the change.
Configuring Audio Features
Acoustic Echo Cancellation (AEC) is used to remove acoustic echo from a voice communication in order to improve the voice quality. It also increases the capacity achieved through silence suppression by preventing echo from traveling across a network. IP phones employ advanced AEC for hands-free operation. Echo cancellation is achieved using the echo canceller.
Procedure
AEC can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure AEC.
For more information, refer to
on page
Configure AEC.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load
To configure AEC via web user interface:
1. Click on Settings->Voice.
2. Select the desired value from the pull-down list of ECHO.
3. Click Confirm to accept the change.
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Voice Activity Detection (VAD) is used in speech processing to detect the presence or absence of human speech. When detecting period of “silence”, VAD replaces that silence efficiently with special packets that indicate silence is occurring. It can facilitate speech processing, and deactivate some processes during non-speech section of an audio session. VAD can avoid unnecessary coding or transmission of silence packets in
VoIP applications, saving on computation and network bandwidth.
Procedure
VAD can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure VAD.
For more information, refer to
on
Configure VAD.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load
To configure VAD via web user interface:
1. Click on Settings->Voice.
2. Select the desired value from the pull-down list of VAD.
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3. Click Confirm to accept the change.
Configuring Audio Features
Comfort Noise Generation (CNG) is used to generate background noise for voice communications during periods of silence in a conversation. It is a part of the silence suppression or VAD handling for VoIP technology. CNG, in conjunction with VAD algorithms, quickly responds when periods of silence occur and inserts artificial noise until voice activity resumes. The insertion of artificial noise gives the illusion of a constant transmission stream, so that background sound is consistent throughout the call and the listener does not think the line has released. The purpose of VAD and CNG is to maintain an acceptable perceived QoS while simultaneously keeping transmission costs and bandwidth usage as low as possible.
Procedure
CNG can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure CNG.
For more information, refer to
on page
Configure CNG.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load
To configure CNG via web user interface:
1. Click on Settings->Voice.
2. Select the desired value from the pull-down list of CNG.
3. Click Confirm to accept the change.
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Jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in even intervals. Jitter is a term indicating variations in packet arrival time, which can occur because of network congestion, timing drift or route changes. The jitter buffer, located at the receiving end of the voice connection, intentionally delays the arriving packets so that the end user experiences a clear connection with very little sound distortion. IP phones support two types of jitter buffers: static and dynamic. A static jitter buffer adds the fixed delay to voice packets. You can configure the delay time for the static jitter buffer on IP phones. A dynamic jitter buffer is capable of adapting the changes in the network's delay. The range of the delay time for the dynamic jitter buffer added to packets can be also configured on IP phones.
Procedure
Jitter buffer can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the mode of jitter buffer and the delay time for jitter buffer.
For more information, refer to
Configure the mode of jitter buffer and the delay time for jitter buffer.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load
To configure Jitter Buffer via web user interface:
1. Click on Settings->Voice.
2. Mark the desired radio box in the Type field.
3. Enter the minimum delay time for adaptive jitter buffer in the Min Delay field.
Valid values range from 0 to 300.
4. Enter the maximum delay time for adaptive jitter buffer in the Max Delay field.
Valid values range from 0 to 300.
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5. Enter the fixed delay time for fixed jitter buffer in the Normal field.
Valid values range from 0 to 300.
6. Click Confirm to accept the change.
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Configuring Security Features
This chapter provides information for making configuration changes for the following security-related features:
Secure Real-Time Transport Protocol
Encrypting Configuration Files
TLS is a commonly-used protocol for providing communications privacy and managing the security of message transmission, allowing IP phones to communicate with other remote parties and connect to the HTTPS URL for provisioning in a way that is designed to prevent eavesdropping and tampering.
TLS protocol is composed of two layers: TLS Record Protocol and TLS Handshake
Protocol. The TLS Record Protocol completes the actual data transmission and ensures the integrity and privacy of the data. The TLS Handshake Protocol allows the server and client to authenticate each other and negotiate an encryption algorithm and cryptographic keys before data is exchanged.
The TLS protocol uses asymmetric encryption for authentication of key exchange, symmetric encryption for confidentiality, and message authentication codes for integrity.
Symmetric encryption
:
For symmetric encryption, the encryption key and the corresponding decryption key can be told by each other. In most cases, the encryption key is the same as the decryption key.
Asymmetric encryption: For asymmetric encryption, each user has a pair of cryptographic keys – a public encryption key and a private decryption key. The information encrypted by the public key can only be decrypted by the corresponding private key and vice versa. Usually, the receiver keeps its private key. The public key is known by the sender, so the sender sends the information encrypted by the known public key, and then the receiver uses the private key to decrypt it.
IP phones support TLS version 1.0. A cipher suite is a named combination of authentication, encryption, and message authentication code (MAC) algorithms used to negotiate the security settings for a network connection using the TLS/SSL network protocol. IP phones support the following cipher suites:
DHE-RSA-AES256-SHA
DHE-DSS-AES256-SHA
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AES256-SHA
EDH-RSA-DES-CBC3-SHA
EDH-DSS-DES-CBC3-SHA
DES-CBC3-SHA
DHE-RSA-AES128-SHA
DHE-DSS-AES128-SHA
AES128-SHA
IDEA-CBC-SHA
DHE-DSS-RC4-SHA
RC4-SHA
RC4-MD5
EXP1024-DHE-DSS-DES-CBC-SHA
EXP1024-DES-CBC-SHA
EDH-RSA-DES-CBC-SHA
EDH-DSS-DES-CBC-SHA
DES-CBC-SHA
EXP1024-DHE-DSS-RC4-SHA
EXP1024-RC4-SHA
EXP1024-RC4-MD5
EXP-EDH-RSA-DES-CBC-SHA
EXP-EDH-DSS-DES-CBC-SHA
EXP-DES-CBC-SHA
EXP-RC4-MD5
The following figure illustrates the TLS messages exchanged between the IP phone and
TLS server to establish an encrypted communication channel:
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Step1: IP phone sends “Client Hello” message proposing SSL options.
Step2: Server responds with “Server Hello” message selecting the SSL options, sends its public key information in “Server Key Exchange” message and concludes its part of the
Configuring Security Features negotiation with “Server Hello Done” message.
Step3: IP phone sends session key information (encrypted by server’s public key) in the
“Client Key Exchange” message.
Step4: Server sends “Change Cipher Spec” message to activate the negotiated options for all future messages it will send.
IP phones can encrypt SIP with TLS, which is called SIPS. When TLS is enabled for an account, the SIP message of this account will be encrypted, and a lock icon appears on the LCD screen after the successful TLS negotiation.
Certificates
The IP phone can serve as a TLS client or a TLS server. The TLS requires the following security certificates to perform the TLS handshake:
Trusted Certificate: When the IP phone requests a TLS connection with a server, the
IP phone should verify the certificate sent by the server to decide whether it is trusted based on the trusted certificates list. The IP phone has 30 built-in trusted certificates. You can upload 10 custom certificates at most. The format of the trusted certificate files must be *.pem,*.cer,*.crt and *.der.
Server Certificate: When clients request a TLS connection with the IP phone, the IP phone sends the server certificate to the clients for authentication. The IP phone has two types of built-in server certificates: a unique server certificate and a generic server certificate. You can only upload one server certificate to the IP phone. The old server certificate will be overridden by the new one. The format of the server certificate files must be *.pem and *.cer.
- A unique server certificate: It is unique to an IP phone (based on the MAC address) and issued by the Yealink Certificate Authority (CA).
- A generic server certificate: It issued by the Yealink Certificate Authority (CA).
Only if no unique certificate exists, the IP phone may send a generic certificate for authentication.
The IP phone can authenticate the server certificate based on the trusted certificates list.
The trusted certificates list and the server certificates list contain the default and custom certificates. You can specify the type of certificates the IP phone accepts: default certificates, custom certificates or all certificates.
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Common Name Validation feature enables the IP phone to mandatorily validate the common name of the certificate sent by the connecting server.
Note In TLS feature, we use the terms trusted and server certificate. These are also known as
CA and device certificates.
Firmware upgrade from version 71 to 72 will result in update of the default server certificates.
We strongly recommend that you do not downgrade the firmware. For
SIP-T20P/T22P/T26P/T28P IP phones, firmware downgrade will result in damage to SSL certificates.
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Configure TLS on a per-line basis.
For more information, refer to
Configure trusted certificates feature.
Configure server certificates feature.
For more information, refer to
Upload the trusted certificates.
Upload the server certificates.
For more information, refer to
Uploading Certificates on page
Configure TLS on a per-line basis.
Navigate to: http://<phoneIPAddress>/servl et?p=account-register&q=load
&acc=0
Configure trusted certificates feature.
Upload the trusted certificates.
Navigate to: http://<phoneIPAddress>/servl
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Configuring Security Features et?p=trusted-cert&q=load
Configure server certificates feature.
Upload the server certificates.
Navigate to: http://<phoneIPAddress>/servl et?p=server-cert&q=load
To configure TLS on a per-line basis via web user interface:
1. Click on Account->Register.
2. Select the desired account from the pull-down list of Account.
3. Select TLS from the pull-down list of Transport.
4. Click Confirm to accept the change.
To configure the trusted certificates via web user interface:
1. Click on Security->Trusted Certificates.
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2. Select the desired values from the pull-down lists of Only Accept Trusted
Certificates, Common Name Validation and CA Certificates.
3. Click Confirm to accept the change.
To upload a trusted certificate via web user interface:
1. Click on Security->Trusted Certificates.
2. Click Browse to select the certificate (*.pem, *.crt, *.cer or *.der) from your local system.
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3. Click Upload to upload the certificate.
To configure the server certificates via web user interface:
1. Click on Security->Server Certificates.
Configuring Security Features
2. Select the desired value from the pull-down list of Device Certificates.
3. Click Confirm to accept the change.
To upload a server certificate via web user interface:
1. Click on Security->Server Certificates.
2. Click Browse to select the certificate (*.pem and *.cer) from your local system.
3. Click Upload to upload the certificate.
A dialog box pops up to prompt “Success: The Server Certificate has been loaded!
Rebooting, please wait…”.
Secure Real-Time Transport Protocol (SRTP) encrypts the RTP streams during VoIP phone calls to avoid interception and eavesdropping. The parties participating in the call must enable SRTP feature simultaneously . When this feature is enabled on both phones, the type of encryption to utilize for the session is negotiated between the IP phones. This negotiation process is compliant with RFC 4568.
When a user places a call on the enabled SRTP phone, the IP phone sends an INVITE message with the RTP encryption algorithm to the destination phone.
Example of the RTP encryption algorithm carried in the SDP of the INVITE message: m=audio 11780 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NzFlNTUwZDk2OGVlOTc3YzNkYTkwZWVkMTM1YWFj
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The callee receives the INVITE message with the RTP encryption algorithm, and then answers the call by responding with a 200 OK message which carries the negotiated
RTP encryption algorithm.
Example of the RTP encryption algorithm carried in the SDP of the 200 OK message: m=audio 11780 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NGY4OGViMDYzZjQzYTNiOTNkOWRiYzRlMjM0Yzcz a=sendrecv a=ptime:20 a=fmtp:101 0-15
SRTP is configurable on a per-line basis. When SRTP is enabled on both IP phones, RTP streams will be encrypted, and a lock icon appears on the LCD screen of each IP phone after successful negotiation.
Note If you enable SRTP, then you should also enable TLS. This ensures the security of SRTP
encryption. For more information on TLS, refer to Transport Layer Security on page 211 .
Procedure
SRTP can be configured using the configuration files or locally.
Configuration File <MAC>.cfg
Configure SRTP feature on a per-line basis.
For more information, refer to
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Configuring Security Features
Local Web User Interface
Configure SRTP feature on a per-line basis.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
To configure SRTP feature via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Advanced.
4. Select the desired value from the pull-down list of RTP Encryption (SRTP).
5. Click Confirm to accept the change.
Encrypted configuration files can be downloaded from the provisioning server to protect against unauthorized access and tampering of sensitive information (e.g., login passwords, registration information). Yealink provides a configuration encryption tool for encrypting configuration files. The encryption tool encrypts plaintext
<y0000000000xx>.cfg and <MAC>.cfg files (one by one or in batch) using 16-character
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For a Microsoft Windows platform, you can use a Yealink-supplied encryption tool
"Config_Encrypt_Tool.exe" to encrypt the <y0000000000xx>.cfg and <MAC>.cfg files respectively.
Note Yealink also provides a configuration encryption tool (yealinkencrypt) for Linux platform if required. For more information, refer to Yealink Configuration Encryption Tool User
Guide .
For security reasons, administrator should upload encrypted configuration files,
<y0000000000xx_Security>.enc
and/or <MAC_Security>.enc files to the root directory of the provisioning server. During auto provisioning, the IP phone requests to download
<y0000000000xx>.cfg file first. If the downloaded configuration file is encrypted, the phone will request to download <y0000000000xx_Security>.enc file (if enabled) and decrypt it into the plaintext key (e.g., key2) using the built-in key (e.g., key1). Then the IP phone decrypts <y0000000000xx>.cfg file using key2. After decryption, the IP phone resolves configuration files and updates configuration settings onto the IP phone system.
The way the IP phone processes the <MAC>.cfg file is the same to that of the<y0000000000xx>.cfg file.
Procedure to Encrypt Configuration Files
To encrypt the <y0000000000xx>.cfg file:
1. Double click “Config_Encrypt_Tool.exe” to start the application tool.
The screenshot of the main page is shown as below:
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When you start the application tool, a file folder named “Encrypted” is created
Configuring Security Features automatically in the directory where the application tool is located.
2. Click Browse to locate configuration file(s) (e.g., y000000000000.cfg) from your local system in the Select File(s) field.
To select multiple configuration files, you can select the first file and then press and hold the Ctrl key and select the next files.
3. (Optional.) Click Browse to locate the target directory from your local system in the
Target Directory field.
The tool uses the file folder “Encrypted” as the target directory by default.
4. (Optional.) Mark the desired radio box in the AES Model field.
If you mark the Manual radio box, you can enter an AES key in the AES KEY field or click Re-Generate to generate an AES key in the AES KEY field. The configuration file(s) will be encrypted using the AES key in the AES KEY field.
If you mark the Auto Generate radio box, the configuration file(s) will be encrypted using random AES key. The AES keys of configuration files are different.
Note AES keys must be 16 characters and the supported characters contain: 0 ~ 9, A ~ Z, a ~ z.
5. Click Encrypt to encrypt the configuration file(s).
6. Click OK.
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The target directory will be automatically opened. You can find the encrypted CFG file(s), encrypted key file(s) and an Aeskey.txt file storing plaintext AES key(s).
Procedure
Decryption method can be configured using the configuration files.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the decryption method.
Configure AES keys.
For more information, refer to
on page
Configure AES keys.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-autop&q=load
To configure AES keys via web user interface:
1. Click on Settings->Auto Provision.
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Configuring Security Features
2. Enter the values in the Common AES Key and MAC-Oriented AES Key fields.
AES keys must be 16 characters and the supported characters contain: 0-9, A-Z, a-z.
3. Click Confirm to accept the change.
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Upgrading Firmware
This chapter provides information about upgrading the IP phone firmware. Two methods of firmware upgrade:
Manually, from the local system.
Automatically, from the provisioning server.
The following table lists the associated firmware name for each IP phone model (X is replaced by the actual firmware version).
IP Phone Model
SIP-T28P
SIP-T26P
SIP-T22P
SIP-T21P
SIP-T20P
SIP-T19P
Associated Firmware Name
2.x.x.x.rom
6.x.x.x.rom
7.x.x.x.rom
34.x.x.x.rom
9.x.x.x.rom
31.x.x.x.rom
Note You can download the latest firmware online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
Upgrade via Web User Interface
To manually upgrade firmware via web user interface, you need to store firmware to your local system in advance.
To upgrade firmware manually via web user interface:
1. Click on Settings->Upgrade.
2. Click Browse.
3. Select firmware from the local system.
4. Click Upgrade.
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A dialog box pops up to prompt “Firmware of the SIP Phone will be updated. It will take
5 minutes to complete. Please don't power off!”.
5. Click OK to confirm the upgrade.
Note Do not unplug the network and power cables when the IP phone is upgrading firmware.
Do not close and refresh the browser when the IP phone is upgrading firmware via web user interface.
Upgrade Firmware from the Provisioning Server
IP phones support to use FTP, TFTP, HTTP, and HTTPS protocols to download the configuration files and firmware from the provisioning server, and then upgrade firmware automatically.
IP phones can download firmware stored on the provisioning server in one of two ways:
Check for both configuration files and firmware stored on the provisioning server during startup.
Automatically check for configuration files and firmware at a fixed interval or specific time.
Method of checking for configuration files and firmware is configurable.
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Configure the way for the IP phone to check for configuration files.
Specify the access URL of firmware.
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Upgrading Firmware
Local Web User Interface
For more information, refer to
on page
Configure the way for the IP phone to check for configuration files.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-autop&q=load
To configure the way for the IP phone to check for new configuration files via web user interface:
1. Click on Settings->Auto Provision.
2. Make the desired change.
3. Click Confirm to accept the change.
When the “Power On” is set to On, the IP phone will check configuration files stored on the provisioning server during startup and then will download firmware from the server.
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Resource Files
When configuring particular features, you may need to upload resource files (e.g., local contact directory, remote phone book) to IP phones. The resources files can be local contact directory, remote phone book and so on. Ask Yealink field application engineer for resource file templates. If the resource file is to be used for all IP phones of the same model, the resource file access URL is best specified in the <y0000000000xx>.cfg file.
However, if you want to specify the desired phone to use the resource file, the resource file access URL should be specified in the <MAC>.cfg file.
This chapter provides the detailed information on how to customize the following resource files and specify the access URL:
Specifying the Access URL of Resource Files
The replace rule template helps with the creation of multiple replace rules. After setup, place the replace rule template to the provisioning server and specify the access URL in the configuration files.
When editing a replace rule template, learn the following:
<DialRule> indicates the start of a template and </DialRule> indicates the end of a template.
Create replace rules between <DialRule> and </DialRule>.
When specifying the desired line(s) to apply the replace rule, the valid values are 0 and line ID. The digit 0 stands for all lines. Multiple line IDs are separated by commas. This is not applicable to SIP-T19P IP phones.
At most 100 replace rules can be added to the IP phone.
The expression syntax in the replace rule template is the same as that introduced
in the section Creating Dial Plan on page 32 .
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Procedure
Use the following procedures to customize a replace rule template.
To customize a replace rule template:
1. Open the template file using an ASCII editor.
2. Add the following string to the template, each starting on a separate line:
<Data Prefix="" Replace="" LineID=""/>
Where:
Prefix="" specifies the numbers to be replaced.
Replace="" specifies the alternate string instead of what the user enters.
LineID="" specifies the desired line(s) for this rule. When you leave it blank or enter
0, this replace rule will apply to all lines.
3. Specify the values within double quotes.
4. Place this file to the provisioning server.
The following is an example of a replace rule template:
<DialRule>
<Data Prefix="1" Replace="05928665234" LineID=""/>
<Data Prefix="2(xx)" Replace="002$1" LineID="0"/>
<Data Prefix="5([6-9])(.)" Replace="3$2" LineID="1,2,3"/>
<Data Prefix="0(.)" Replace="9$1" LineID="2"/>
<Data Prefix="1009" Replace="05921009" LineID="1"/>
</DialRule>
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The dial-now template helps with the creation of multiple dial-now rules. After setup, place the dial-now template to the provisioning server and specify the access URL in the configuration files.
When editing a dial-now template, learn the following:
<DialNow> indicates the start of a template and </DialNow> indicates the end of a template.
Create dial-now rules between <DialNow> and </DialNow>.
When specifying the desired line(s) for the dial-now rule, the valid values are 0 and line ID. 0 stands for all lines. Multiple line IDs are separated by commas. This is not applicable to SIP-T19P IP phones.
At most 100 rules can be added to the IP phone.
The expression syntax in the dial-now rule template is the same as that introduced
Resource Files
in the section Creating Dial Plan on page 32 .
Procedure
Use the following procedures to customize a dial-now template.
To customize a dial-now template:
1. Open the template file using an ASCII editor.
2. Add the following string to the template, each starting on a separate line:
<Data DialNowRule="" LineID=""/>
Where:
DialNowRule="" specifies the dial-now rule.
LineID="" specifies the desired line(s) for this rule. When you leave it blank or enter
0, this dial-now rule will apply to all lines.
3. Specify the values within double quotes.
4. Place this file to the provisioning server.
The following is an example of a dial-now template:
<DialNow>
<Data DialNowRule="1234" LineID="1"/>
<Data DialNowRule="52[0-6]" LineID="1"/>
<Data DialNowRule="xxxxxx" LineID=""/>
</DialNow>
The softkey layout template allows you to assign different soft key layouts to different call states. The call states include CallFailed, CallIn, Connecting, Dialing, RingBack and
Talking. After setup, place the templates to the provisioning server and specify the access URL in the configuration files.
When editing a softkey layout template, learn the following:
<Call States> indicates the start of a template and </Call States> indicates the end of a template. For example, <CallFailed></CallFailed>.
<Disable> indicates the start of the disabled soft key list and </Disable> indicates the end of the soft key list, the disabled soft keys are not displayed on the LCD screen.
Create disabled soft keys between <Disable> and </Disable>.
<Enable> indicates the start of the enabled soft key list and </Enable> indicates the end of the soft key list, the enabled soft keys are displayed on the LCD screen.
Create enabled soft keys between <Enable> and </Enable>.
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<Default> indicates the start of the default soft key list and </Default> indicates the end of the default soft key list, the default soft keys are displayed on the LCD screen by default.
Procedure
Use the following procedures to customize a softkey layout template.
To customize a softkey layout template:
1. Open the template file using an ASCII editor.
2. For each soft key that you want to enable, add the following string to the file. Each starts on a separate line:
<Key Type=""/>
Where:
Key Type="" specifies the enabled soft key (This value cannot be blank).
For each disabled soft key and each default soft key that you want to add, add the same string introduced above.
3. Specify the values within double quotes.
4. Place this file to the provisioning server.
The following is an example of the CallFailed template:
<CallFailed>
<Disable>
<Key Type="Empty"/>
<Key Type="Switch"/>
<Key Type="Cancel"/>
</Disable>
<Enable>
<Key Type="NewCall"/>
<Key Type="Empty"/>
<Key Type="Empty"/>
<Key Type="Empty"/>
</Enable>
<Default>
<Key Type="NewCall"/>
<Key Type="Empty"/>
<Key Type="Empty"/>
<Key Type="Empty"/>
</Default>
</CallFailed>
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Resource Files
You can add contacts one by one on the IP phone directly. You can also add multiple contacts at a time and/or share contacts between IP phones using the local contact template file. After setup, place the template file to the provisioning server and specify the access URL of the template file in the configuration files.
When editing a local contact template, learn the following:
<root_contact> indicates the start of a contact list and </root_contact> indicates the end of a contact list.
<root_group> indicates the start of a group list and </root_group> indicates the end of a group list.
When specifying a ring tone for a contact or a group, the format of the value must be Auto (the first registered line), Resource:RingN.wav (system ring tone, integer N ranges from 1 to 5) or Custom:Name.wav (custom ring tone).
When specifying a desired line for a contact, the valid values are 0 and line ID, 0 stands for the first available account. Multiple line IDs are separated by commas.
At most 5 groups can be added to the IP phone.
At most 1000 local contacts can be added to the IP phone.
Procedure
Use the following procedures to customize a local contact template file.
To customize a local contact file:
1. Open the template file using an ASCII editor.
2. For each group that you want to add, add the following string to the file. Each starts on a separate line:
<group display_name="" ring=""/>
Where: display_name="" specifies the name of the group. ring="" specifies the desired ring tone for this group.
3. For each contact that you want to add, add the following string to the file. Each starts on a separate line:
<contact display_name="" office_number="" mobile_number="" other_number="" line="" ring="" group_id_name=""/>
Where: display_name="" specifies the name of the contact (This value cannot be blank or duplicated). office_number="" specifies the office number of the contact.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones mobile_number="" specifies the mobile number of the contact. other_number="" specifies the other number of the contact. line="" specifies the line you want to add this contact to. ring="" specifies the ring tone for this contact. group_id_name="" specifies the existing group you want to add the contact to.
4. Specify the values within double quotes.
5. Place this file to the provisioning server.
The following is an example of a local contact file:
<root_group>
<group display_name="Friend" ring=""/>
<group display_name="Family" ring="Resource:Ring1.wav"/>
</root_group>
<root_contact>
<contact display_name="John" office_number="1001" mobile_number="12345678910" other_number="" line="0" ring="Auto" group_id_name="All Contacts"/>
<contact display_name="Alice" office_number="1002" mobile_number="" other_number="" line="1,2" ring=”Resource:Ring2.wav” group_id_name="Friend"/>
</root_contact>
234
IP phones can access 5 remote phone books. You can customize the remote XML phone book for IP phones as required. Before specifying the access URL of the remote phone book in the configuration files, you need to create a remote XML phone book and then place it to the provisioning server.
When creating an XML phone book, learn the following:
<YealinkIPPhoneDirectory> indicates the start of a phone book and
</YealinkIPPhoneDirectory> indicates the end of a phone book.
<DirectoryEntry> indicates the start of a contact and </DirectoryEntry> indicates the end of a contact.
Procedure
Use the following procedures to customize an XML phone book.
Customizing an XML phone book:
1. Open the template file using an ASCII editor.
Resource Files
2. For each contact that you want to add, add the following strings to the phone book.
Each starts on a separate line:
<Name>
Mary
</Name>
<Telephone>
1001
</Telephone>
Where:
Specify the contact name between <Name> and </Name>.
Specify the contact number between <Telephone> and </Telephone>.
3. Specify the values within double quotes.
4. Place this file to the provisioning server.
The following is an example of an XML phone book:
<YealinkIPPhoneDirectory>
<DirectoryEntry>
<Name>Jack</Name>
<Telephone>1003</Telephone>
</DirectoryEntry>
<DirectoryEntry>
<Name>John</Name>
<Telephone>1004</Telephone>
</DirectoryEntry>
<DirectoryEntry>
<Name>Marry</Name>
<Telephone>1005</Telephone>
</DirectoryEntry>
</YealinkIPPhoneDirectory>
Note Yealink supplies a phonebook generation tool to generate a remote XML phone book.
For more information, refer to Yealink Phonebook Generation Tool User Guide .
Directory provides easy access to frequently used lists. The lists may contain Local
Directory, History, Remote Phone Book and LDAP. Users can access the lists by pressing the Directory soft key when the IP phone is idle. After setup, place the directory template to the provisioning server and specify the access URL in the configuration files.
When editing a directory template, learn the following:
<root_favorite_set> indicates the start of a template and </root_favorite_set> indicates the end of a template.
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Create directory between <root_favorite_set> and </root_favorite_set>.
When specifying the display name of the directory list, the valid values are Local
Contacts, History, Remote Phone Book (not applicable to SIP-T20P IP phones) and
LDAP (not applicable to SIP-T19P and SIP-T20P IP phones).
When specifying the display priority of the directory list, the valid values are 1, 2, 3 and 4. 1 is the highest priority, 4 is the lowest.
When enabling or disabling the desired directory list, the valid values are 0 and 1.
0 stands for Disabled, 1 stands for Enabled.
Procedure
Use the following procedures to customize a directory template.
Customizing a directory template:
1. Open the template file using an ASCII editor.
2. For each directory list that you want to configure, add the following string to the file.
Each starts on a separate line:
<item id_name="" display_name="" priority="" enable="" />
Where: id_name="" specifies the existing directory list you want to configure. We do not recommend editing this field. display_name="" specifies the display name of the directory list. We do not recommend editing this field. priority="" specifies the display priority of the directory list. enable="" enables or disables the directory list.
3. Specify the values within double quotes.
4. Place this file to the provisioning server.
The following is an example of a directory template:
<root_favorite_set>
<item id_name="localdirectory" display_name="Local Directory" priority="1" enable="1" />
<item id_name="history" display_name="History" priority="2" enable="0" />
<item id_name="remotedirectory" display_name="Remote Phone Book" priority="3" enable="0" />
<item id_name="ldap" display_name="LDAP" priority="4" enable="0" />
</root_favorite_set>
236
Resource Files
The super search template allows you to search for a contact in your desired lists when the phone is in the dialing screen. The lists may contain Local Directory, History, Remote
Phone Book and LDAP. After setup, place the super search template to the provisioning server and specify the access URL in the configuration files.
When editing a super search template, learn the following:
<root_super_search> indicates the start of a template and </root_super_search> indicates the end of a template.
Create super search between <root_super_search> and </root_super_search>.
When specifying the display name of the directory list, the valid values are Local
Contacts, History, Remote Phone Book (not applicable to SIP-T20P IP phones) and
LDAP (not applicable to SIP-T19P and SIP-T20P IP phones).
When specifying the priority of search results, the valid values are 1, 2, 3 and 4. 1 is the highest priority, 4 is the lowest.
When enabling or disabling the desired directory list, the valid values are 0 and 1.
0 stands for Disabled, 1 stands for Enabled.
Procedure
Use the following procedures to customize a super search template.
Customizing a super search template:
1. Open the template file using an ASCII editor.
2. For each directory list that you want to configure, add the following string to the file.
Each starts on a separate line:
<item id_name="" display_name="" priority="" enable="" />
Where: id_name="" specifies the existing directory list you want to configure. We do not recommend editing this field. display_name="" specifies the display name of the directory list. We do not recommend editing this field. priority="" specifies the priority of search results. enable="" enables or disables the directory list.
3. Specify the values within double quotes.
4. Place this file to the provisioning server.
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The following is an example of a super search template:
<root_super_search>
<item id_name="local_directory_search" display_name="Local Contacts" priority="1" enable="1" />
<item id_name="calllog_search" display_name="History" priority="2" enable="1" />
<item id_name="remote_directory_search" display_name="Remote Phone
Book" priority="3" enable="0" />
<item id_name="ldap_search" display_name="LDAP" priority="4" enable="0" />
</root_super_search>
238
Access URL of the resource file can be configured in the configuration files:
Configuration File
Configuration File
Configuration File
Configuration File
Configuration File
<y0000000000xx>.cfg
<y0000000000xx>.cfg
<y0000000000xx>.cfg
<y0000000000xx>.cfg
<y0000000000xx>.cfg
Configure the access URL of the replace rule template.
For more information, refer to
Configure the access URL of the dial-now rule template.
For more information, refer to
on page
Configure the access URL of the softkey layout template.
For more information, refer to
Configure the access URL of the local contact file.
For more information, refer to
Configure the access URL of the remote XML phone book.
For more information, refer to
Resource Files
Configuration File
Configuration File
<y0000000000xx>.cfg
<y0000000000xx>.cfg
Configure the access URL of the directory template.
For more information, refer to
Configure the access URL of the super search template.
For more information, refer to
on page
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
240
Troubleshooting
This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using IP phones.
IP phones can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which can help an administrator more easily find the system problem and fix it.
The following are helpful for better understanding and resolving the working status of the IP phone.
Getting Information from Status Indicators
If your IP phone encounters some problems, commonly the log files are used. You can export the log files to a syslog server or the local system. You can also specify the severity level of the log to be reported to a log file. The default system log level is 3
(Changes to this parameter via web user interface require a reboot).
In the configuration files, you can use the following parameters to configure system log settings:
syslog.server -- Specify the IP address of the syslog server to which the log will be exported. syslog.log_level -- Specify the system log level.
For more information on the system log setting parameters, refer to Log Settings on page 397 .
To configure the level of the system log via web user interface:
1. Click on Settings->Configuration.
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2. Select the desired level from the pull-down list of System Log Level.
3. Click Confirm to accept the change.
A dialog box pops up to prompt “Do you want to restart your machine?”. The configuration will take effect after a reboot.
4. Click OK to reboot the IP phone.
After a reboot, the system log level is set as 6, the administrator debug level.
Note Administrator level debugging may make some sensitive information accessible (e.g., password-dial number), we recommend that you reset the system log level to 3 after having the syslog file provided.
To configure the phone to export the system log to a syslog server via web user interface:
1. Click on Settings->Configuration.
2. Mark the Server radio box in the Export System Log field.
3. Enter the IP address or domain name of the syslog server in the Server Name field.
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Troubleshooting
4. Click Confirm to accept the change.
A dialog box pops up to prompt “Do you want to restart your machine?”. The configuration will take effect after a reboot.
5. Click OK to reboot the IP phone.
The system log will be exported successfully to the desired syslog server after a reboot.
6. Reproduce the issue.
To export a log file to the local system via web user interface:
1. Click on Settings->Configuration.
2. Mark the Local radio box in the Export System Log field.
3. Reproduce the issue.
4. Click Export to open file download window, and then save the file to your local system.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
The following figure shows a portion of a log file:
You can capture packet in two ways: capturing the packet via web user interface or using the Ethernet software. You can analyze the packet captured for troubleshooting purpose.
To capture packets via web user interface:
1. Click on Settings->Configuration.
2. Click Start to start capturing signal traffic.
3. Reproduce the issue to get stack traces.
4. Click Stop to stop capturing.
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Troubleshooting
5. Click Export to open the file download window, and then save the file to your local system.
To capture packets using the Ethernet software:
Connect the Internet port of the IP phone and the PC to the same HUB, and then use
Sniffer, Ethereal or Wireshark software to capture the signal traffic.
The IP phone provides a troubleshooting feature called “Watch Dog”, which helps you monitor the IP phone status and provides the ability to get stack traces from the last time the IP phone failed. If Watch Dog feature is enabled, the IP phone will automatically reboot when it detects a fatal failure. This feature can be configured using the configuration files or via web user interface.
You can use the “watch_dog.enable” parameter to configure watch dog feature in the
configuration files. For more information, refer to Watch Dog on page 398 .
To configure watch dog feature via web user interface:
1. Click on Settings->Preference.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
2. Select the desired value from the pull-down list of Watch Dog.
3. Click Confirm to accept the change.
Status indicators may consist of the power LED, MESSAGE key LED, line key indicator, headset key indicator and the on-screen icon.
The following shows two examples of obtaining the phone information from status indicators:
If a LINK failure of the IP phone is detected, a prompting message “Network
Unavailable” and the icon will appear on the LCD screen.
If a voice mail is received, the MESSAGE key LED illuminates.
For more information on the icons, refer to
Wrong configurations may have an impact on your phone use. You can export configuration file to check the current configuration of the IP phone and troubleshoot if necessary.
To export configuration file via web user interface:
1. Click on Settings->Configuration.
246
Troubleshooting
2. In the Export or Import Configuration block, click Export to open the file download window, and then save the file to your local system.
This section describes solutions to common issues that may occur while using the IP phone. Upon encountering a scenario not listed in this section, contact your Yealink reseller for further support.
Do one of the following:
Ensure that the IP phone is properly plugged into a functional AC outlet.
Ensure that the IP phone is plugged into a socket controlled by a switch that is on.
If the IP phone is plugged into a power strip, try plugging it directly into a wall outlet.
If your phone is PoE powered, ensure that you are using a PoE-compliant switch or hub.
’
Do one of the following:
Ensure that the Ethernet cable is plugged into the Internet port on the IP phone and the Ethernet cable is not loose.
Ensure that the Ethernet cable is not damaged.
Ensure that the IP address and related network parameters are set correctly.
Ensure that your network switch or hub is operational.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
“ ”
The LCD screen prompts “No Service” message when there is no available SIP account on the IP phone.
Do one of the following:
Ensure that an account is actively registered on the IP phone at the path
Menu->Status->More->Accounts.
Ensure that the SIP account parameters have been configured correctly.
Press the OK key when the IP phone is idle to check the basic information (e.g., IP address, MAC address and firmware version).
’
Do one of the following:
Ensure that the target firmware is not the same as the current firmware.
Ensure that the target firmware is applicable to the IP phone model.
Ensure that the current or the target firmware is not protected.
Ensure that the power is on and the network is available in the process of upgrading.
Ensure that the web browser is not closed or refreshed when upgrading firmware via web user interface.
’
Check if the IP phone is configured to obtain the time and date from the NTP server automatically. If your phone is unable to access the NTP server, configure the time and date manually.
248
If you have poor sound quality/acoustics like intermittent voice, low volume, echo or other noises, the possible reasons could be:
Users are seated too far out of recommended microphone range and sound faint, or are seated too close to sensitive microphones and cause echo.
Intermittent voice is mainly caused by packet loss, due to network congestion, and
Troubleshooting
jitter, due to message recombination of transmission or receiving equipment (e.g., timeout handling, retransmission mechanism, buffer under run).
Noisy equipment, such as a computer or a fan, may cause voice interference. Turn off any noisy equipment.
Line issues can also cause this problem; disconnect the old line and redial the call to ensure another line may provide better connection.
A remote phone book is placed on a server, while a local phone book is placed on the
IP phone flash. A remote phone book can be used by everyone that can access the server, while a local phone book can only be used by a specific phone. A remote phone book is always used as a central phone book for a company; each employee can load it to obtain the real-time data from the same server.
Both user name and register name are defined by the server. User name identifies the account, while register name matched with a password is for authentication purposes.
Display name is the caller ID that will be displayed on the callee’s phone LCD screen.
Server configurations may override the local ones.
IP phones support remote reboot by a SIP NOTIFY message with “Event: check-sync” header. When receiving a NOTIFY message with the parameter “reboot=true”, the IP phone reboots immediately. The NOTIFY message is formed as shown:
NOTIFY sip:<user>@<dsthost> SIP/2.0
To: sip:<user>@<dsthost>
From: sip:sipsak@<srchost>
CSeq: 10 NOTIFY
Call-ID: 1234@<srchost>
Event: check-sync;reboot=true
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
The IP phone only uses logo file in DOB format, as the DOB format file has a high compression ratio (the size of the uncompressed file compared to that of the compressed file) and can be stored in smaller space. Tools for converting BMP format to
DOB format are available. For more information, refer to
Yealink_SIP-T2
Series_T19P_T4_Series_IP_Phones_Auto_Provisioning_Guide
.
Press the volume key to increase or decrease the ringer volume when the phone is idle, or to adjust the volume of engaged audio device (handset, speakerphone or headset) when there is an active call in progress.
IP phones manufactured before February 2010 will use the power adapter preferentially, while those made later will use PoE preferentially.
Auto provisioning refers to the update of IP phones, including update on configuration parameters, local phone book, firmware and so on. You can use auto provisioning on a single phone, but it makes more sense in mass deployment.
Plug and Play (PnP) is a method for IP phones to acquire the provisioning server address.
With PnP enabled, the IP phone broadcasts the PnP SUBCRIBE message to obtain a provisioning server address during startup. Any SIP server recognizing the message will respond with the preconfigured provisioning server address, so the IP phone will be able to download the CFG files from the provisioning server. PnP depends on support from a SIP server.
250
Troubleshooting
’
Do one of the following:
Ensure that the configuration is set correctly.
Reboot the IP phone. Some configurations require a reboot to take effect.
Ensure that the configuration is applicable to the IP phone model.
The configuration may depend on support from a server.
“ ” “ ”
They are codes that the IP phone sends to the server when a certain action takes place.
On code is used to activate a feature on the server side, while off code is used to deactivate a feature on the server side.
For example, if you set the Always Forward on code to be *78 (may vary on different servers), and the target number to be 201. When you enable Always Forward on the IP phone, the IP phone sends *78201 to the server, and then the server will enable Always
Forward feature on the server side, hence being able to get the right status of the extension.
Do one of the following:
Reset another available IP address for the IP phone.
Check network configuration via phone user interface at the path
Menu->Settings->Advanced Settings->Network->WAN Port->IPv4. If Static IP
Client is selected, select DHCP IP Client instead.
Reset your phone to factory configurations after you have tried all troubleshooting suggestions but do not solve the problem. Note that all custom settings will be overwritten after resetting.
To reset the IP phone via web user interface:
1. Click on Settings->Upgrade.
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2. Click Reset to Factory Reset in the Reset to Factory Setting field.
The web user interface prompts the message “Do you want to reset to factory?”.
3. Click OK to confirm the resetting.
The phone will be reset to factory sucessfully after startup.
Note Reset of your phone may take a few minutes. Do not power off until the phone starts up successfully.
Factory reset can restore the original password. All custom settings will be overwritten after reset.
252
Phone Model
SIP-T28P
SIP-T26P
SIP-T22P
LCD
320*160 pixel
132*64 pixel
132*64 pixel
Logo
Display
236*82 pixel
132*64 pixel
132*64 pixel
Line
Key
6
3
3
Memory
Key
10
10
/
SMS
XML
Browser
Support Support
Support Support
Support Support
Troubleshooting
Phone Model
SIP-T21P
SIP-T20P
SIP-T19P
LCD
Logo
Display
132*64 pixel
3-line
(2*15 characte rs and an icon line)
132*64 pixel
132*64 pixel
Text log
132*64 pixel
Line
Key
2
2
/
Memory
Key
/
/
/
SMS
XML
Browser
Support Support
/
Support
(Non UI)
Support Support
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254
Appendix
802.1x--an IEEE Standard for port-based Network Access Control (PNAC). It is a part of the IEEE 802.1 group of networking protocols. It provides an authentication mechanism to devices wishing to attach to a LAN or WLAN.
ACD (Automatic Call Distribution)--used to distribute calls from large volumes of incoming calls to the registered IP phone users.
ACS (Auto Configuration server)--responsible for auto-configuration of the Central
Processing Element (CPE).
Cryptographic Key--a piece of variable data that is fed as input into a cryptographic algorithm to perform operations such as encryption and decryption, or signing and verification.
DHCP (Dynamic Host Configuration Protocol)--built on a client-server model, where designated DHCP server hosts allocate network addresses and deliver configuration parameters to dynamically configured hosts.
DHCP Option--can be configured for specific values and enabled for assignment and distribution to DHCP clients based on server, scope, class or client-specific levels.
DNS (Domain Name System)--a hierarchical distributed naming system for computers, services, or any resource connected to the Internet or a private network.
EAP-MD5 (Extensible Authentication ProtocolMessage Digest Algorithm 5 )--only provides authentication of the EAP peer to the EAP server but not mutual authentication.
EAP-TLS (Extensible Authentication Protocol-Transport Layer Security) –provides for mutual authentication, integrity-protected cipher suite negotiation between two endpoints.
PEAP-MSCHAPv2 (Protected Extensible Authentication Protocol-Microsoft Challenge
Handshake Authentication Protocol version 2) –provides for mutual authentication, but does not require a client certificate on the IP phone.
FAC (Feature Access Code)--special patterns of characters that are dialed from a phone keypad to invoke particular features.
HTTP (Hypertext Transfer Protocol)--used to request and transmit data on the World
Wide Web.
HTTPS (Hypertext Transfer Protocol over Secure Socket Layer)--a widely-used communications protocol for secure communication over a network.
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IEEE (Institute of Electrical and Electronics Engineers)--a non-profit professional association headquartered in New York City that is dedicated to advancing technological innovation and excellence.
LAN (Local Area Network)--used to interconnects network devices in a limited area such as a home, school, computer laboratory, or office building.
MIB (Management Information Base)--a virtual database used for managing the entities in a communications network.
OID (Object Identifier)--assigned to an individual object within a MIB.
PnP (Plug and Play)--a term used to describe the characteristic of a computer bus, or device specification, which facilitates the discovery of a hardware component in a system, without the need for physical device configuration, or user intervention in resolving resource conflicts.
ROM (Read-only Memory)--a class of storage medium used in computers and other electronic devices.
RTP (Real-time Transport Protocol)--provides end-to-end service for real-time data.
TCP (Transmission Control Protocol)--a transport layer protocol used by applications that require guaranteed delivery.
UDP (User Datagram Protocol)--a protocol offers non-guaranteed datagram delivery.
URI (Uniform Resource Identifier)--a compact sequence of characters that identifies an abstract or physical resource.
URL (Uniform Resource Locator)--specifies the address of an Internet resource.
VLAN (Virtual LAN)-- a group of hosts with a common set of requirements, which communicate as if they were attached to the same broadcast domain, regardless of their physical location.
VoIP (Voice over Internet Protocol)--a family of technologies used for the delivery of voice communications and multimedia sessions over IP networks.
WLAN (Wireless Local Area Network)--a type of local area network that uses high-frequency radio waves rather than wires to communicate between nodes.
XML-RPC (Remote Procedure Call Protocol)--which uses XML to encode its calls and
HTTP as a transport mechanism.
256
Appendix
−04:00
−04:00
−04:00
−04:00
−04:00
−04:00
−03:30
−03:00
−03:00
−03:00
−03:00
−02:00
−01:00
0
0
0
0
0
0
−07:00
−07:00
−07:00
−06:00
−06:00
−06:00
−06:00
−05:00
−05:00
−05:00
−05:00
−04:30
Time Zone
−11:00
−10:00
−10:00
−09:00
−08:00
−08:00
−08:00
−07:00
Time Zone Name
Samoa
United States-Hawaii-Aleutian
United States-Alaska-Aleutian
United States-Alaska Time
Canada(Vancouver, Whitehorse)
Mexico(Tijuana, Mexicali)
United States-Pacific Time
Canada(Edmonton, Calgary)
Mexico(Mazatlan, Chihuahua)
United States-Mountain Time
United States-MST no DST
Canada-Manitoba(Winnipeg)
Chile(Easter Islands)
Mexico(Mexico City, Acapulco)
United States-Central Time
Bahamas(Nassau)
Canada(Montreal, Ottawa, Quebec)
Cuba(Havana)
United States-Eastern Time
Venezuela(Caracas)
Canada(Halifax, Saint John)
Chile(Santiago)
Paraguay(Asuncion)
United Kingdom-Bermuda(Bermuda)
United Kingdom(Falkland Islands)
Trinidad&Tobago
Canada-New Foundland(St.Johns)
Denmark-Greenland(Nuuk)
Argentina(Buenos Aires)
Brazil(no DST)
Brazil(DST)
Brazil(no DST)
Portugal(Azores)
GMT
Greenland
Denmark-Faroe Islands(Torshavn)
Ireland(Dublin)
Portugal(Lisboa, Porto, Funchal)
Spain-Canary Islands(Las Palmas)
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+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+02:00
+02:00
+02:00
+02:00
+02:00
Time Zone
0
0
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+01:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+02:00
+03:00
+03:00
+03:00
+03:30
+04:00
+04:00
+04:00
+04:00
+04:00
Time Zone Name
United Kingdom(London)
Morocco
Albania(Tirane)
Austria(Vienna)
Belgium(Brussels)
Caicos
Chad
Spain(Madrid)
Croatia(Zagreb)
Czech Republic(Prague)
Denmark(Kopenhagen)
France(Paris)
Germany(Berlin)
Hungary(Budapest)
Italy(Rome)
Luxembourg(Luxembourg)
Macedonia(Skopje)
Netherlands(Amsterdam)
Namibia(Windhoek)
Estonia(Tallinn)
Finland(Helsinki)
Gaza Strip(Gaza)
Greece(Athens)
Israel(Tel Aviv)
Jordan(Amman)
Latvia(Riga)
Lebanon(Beirut)
Moldova(Kishinev)
Russia(Kaliningrad)
Romania(Bucharest)
Syria(Damascus)
Turkey(Ankara)
Ukraine(Kyiv, Odessa)
East Africa Time
Iraq(Baghdad)
Russia(Moscow)
Iran(Teheran)
Armenia(Yerevan)
Azerbaijan(Baku)
Georgia(Tbilisi)
Kazakhstan(Aktau)
Russia(Samara)
258
+08:00
+08:00
+09:00
+09:00
+09:30
+09:30
+10:00
+10:00
+10:00
+10:00
+10:30
+11:00
+12:00
+12:45
+13:00
Time Zone
+04:30
+05:00
+05:00
+05:00
+05:00
+05:30
+06:00
+06:00
+07:00
+07:00
+08:00
Appendix
Time Zone Name
Afghanistan
Kazakhstan(Aqtobe)
Kyrgyzstan(Bishkek)
Pakistan(Islamabad)
Russia(Chelyabinsk)
India(Calcutta)
Kazakhstan(Astana, Almaty)
Russia(Novosibirsk, Omsk)
Russia(Krasnoyarsk)
Thailand(Bangkok)
China(Beijing)
Singapore(Singapore)
Australia(Perth)
Korea(Seoul)
Japan(Tokyo)
Australia(Adelaide)
Australia(Darwin)
Australia(Sydney, Melbourne, Canberra)
Australia(Brisbane)
Australia(Hobart)
Russia(Vladivostok)
Australia(Lord Howe Islands)
New Caledonia(Noumea)
New Zealand(Wellington, Auckland)
New Zealand(Chatham Islands)
Tonga(Nukualofa)
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This appendix describes configuration parameters in the configuration files for each feature. The configuration files are <y0000000000xx>.cfg and <MAC>.cfg.
You can set parameters in the configuration files to configure IP phones. The
<y0000000000xx>.cfg and <MAC>.cfg files are stored on the provisioning server. The
IP phone checks for configuration files and looks for resource files when restarting the IP phone. The <y0000000000xx>.cfg file stores configurations for all phones of the same model. The <MAC>.cfg file stores configurations for a specific IP phone with that MAC address.
Configuration changes made in the <MAC>.cfg file override the configuration settings in the <y0000000000xx>.cfg file.
Parameter- network.internet_port.type
Description
Format
Default Value
Range
Example
Parameter- network.static_dns_enable
Description
Configuration File
<MAC>.cfg
Configures the Internet port type.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
Valid values are:
0-DHCP
1-PPPoE
2-Static IP Address network.internet_port.type= 0
Configuration File
<y0000000000xx>.cfg
Enables or disables the phone to use manually configured static IPv4 DNS when the
260
Format
Default Value
Range
Example
Appendix parameter “network.internet_port.type” is set to 0 (DHCP).
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
0
Valid values are:
0-Disabled
1-Enabled network.static_dns_enable= 0
Parameter- network.internet_port.type
Description
Format
Default Value
Range
Example
Parameter- network.ip_address_mode
Configuration File
<MAC>.cfg
Configures the Internet port type.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
Valid values are:
0-DHCP
1-PPPoE
2-Static IP Address network.internet_port.type = 2
Description
Configuration File
<MAC>.cfg
Configures the IP address mode.
IP phones support to use the IPv4 address only, the IPv6 address only or both IPv4 and IPv6 addresses.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Format
Default Value
Range
Integer
0
Valid values are:
0-IPv4
1-IPv6
2-IPv4&IPv6 network.ip_address_mode = 0 Example
Parameter- network.internet_port.ip
Description
Format
Default Value
Range
Example
Parameter- network.internet_port.mask
Configuration File
<MAC>.cfg
Configures the IP address when the Internet port type is configured as Static IP Address and the IP address mode is configured as IPv4 or IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
IPv4 Address
Blank
Not Applicable network.internet_port.ip = 192.168.1.20
Description
Configuration File
<MAC>.cfg
Configures the subnet mask when the Internet port type is configured as Static IP Address and the IP address mode is configured as IPv4 or IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Subnet Mask
Blank
Not Applicable network.internet_port.mask = 255.255.255.0
Format
Default Value
Range
Example
262
Appendix
Parameter- network.internet_port.gateway
Configuration File
<MAC>.cfg
Description
Format
Configures the default gateway when the
Internet port type is configured as Static IP
Address and the IP address mode is configured as IPv4 or IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
IPv4 Address
Default Value
Range
Example
Blank
Not Applicable network.internet_port.gateway =
192.168.1.254
Parameter- network.primary_dns
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the primary DNS server when the
Internet port type is configured as Static IP
Address and the IP address mode is configured as IPv4 or IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
IPv4 Address
Blank
Not Applicable network.primary_dns = 202.101.103.55
Parameter- network.secondary_dns
Description
Configuration File
<MAC>.cfg
Configures the secondary DNS server when the Internet port type is configured as Static IP
Address and the IP address mode is configured as IPv4 or IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take
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Format
Default Value
Range
Example effect.
IPv4 Address
Blank
Not Applicable network.secondary_dns = 202.101.103.54
264
Parameter- network.internet_port.type
Description
Format
Default Value
Range
Configuration File
<MAC>.cfg
Configures the Internet port type.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
Valid values are:
0-DHCP
1-PPPoE
2-Static IP Address network.internet_port.type= 1 Example
Parameter- network.pppoe.user
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the PPPoE user name when the
Internet port type is configured as PPPoE and the IP address mode is configured as IPv4 or
IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
String
Blank
String within 32 characters network.pppoe.user = xmyealink
Parameter- network.pppoe.password
Description
Format
Default Value
Range
Example
Appendix
Configuration File
<y0000000000xx>.cfg
Configures the PPPoE password when the
Internet port type is configured as PPPoE and the IP address mode is configured as IPv4 or
IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
String
Blank
String within 99 characters network.pppoe.password = yealink123
Internet Port Transmission Method
Parameter- network.internet_port.speed_d
uplex
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures the transmission method of Internet port.
Note: We recommend that you do not change this parameter. If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
Valid values are:
0-Auto negotiate
1-Full duplex, 10Mbps
2-Full duplex, 100Mbps
3-Half duplex, 10Mbps
4-Half duplex, 100Mbps network.internet_port.speed_duplex = 0
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PC Port Transmission Method
Parameter- network.pc_port.speed_duplex
Configuration File
<y0000000000xx>.cfg
Description
Configures the transmission method of PC port.
Note: We recommend that you do not change this parameter. If you change this parameter, the IP phone will reboot to make the change take effect.
Format
Default Value
Range
Example
Integer
0
Valid values are:
0-Auto negotiate
1-Full duplex, 10Mbps
2-Full duplex, 100Mbps
3-Half duplex, 10Mbps
4-Half duplex, 100Mbps network.pc_port.speed_duplex = 0
Parameter- network.PC_port.enable
Description
Format
Default Value
Range
Example
Parameter- network.bridge_mode
Description
266
Configuration File
<y0000000000xx>.cfg
Enables or disables the PC port.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
1
Valid values are:
0-Disabled
1-Auto Negotiation network.PC_port.enable = 1
Configuration File
<y0000000000xx>.cfg
Configures the PC port mode.
Format
Default Value
Range
Example
Parameter- network.pc_port.ip
Description
Format
Default Value
Range
Example
Parameter- network.pc_port.mask
Description
Format
Default Value
Range
Appendix
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is not applicable to SIP-T19P and
SIP-T21P IP phones. SIP-T19P and SIP-T21P IP phones only support bridge mode for PC connection.
Integer
1
Valid values are:
0-Router
1-Bridge network.bridge_mode = 1
Configuration File
<y0000000000xx>.cfg
Configures the IP address for the PC port when the PC port is configured as Router.
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is not applicable to SIP-T19P and
SIP-T21P IP phones.
IP Address
10.0.0.1
Not Applicable network.pc_port.ip = 10.0.0.1
Configuration File
<y0000000000xx>.cfg
Configures the subnet mask for the PC port when the PC port is configured as Router.
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is not applicable to SIP-T19P and
SIP-T21P IP phones.
IP Address
255.255.255.0
Not Applicable
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Example network.pc_port.mask = 255.255.255.0
Parameter- network.pc_port.dhcp_server
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the DHCP service for the
PC attached to the PC port when the PC port is configured as Router.
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is not applicable to SIP-T19P and
SIP-T21P IP phones.
Boolean
1
Valid values are:
0-Disabled
1-Enabled network.pc_port.dhcp_server = 1
Parameter- network.dhcp.start_ip
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the start IP address that the IP phone assigns for the PC attached to the PC port when the PC port is configured as Router.
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is not applicable to SIP-T19P and
SIP-T21P IP phones.
IP Address
10.0.0.10
Not Applicable network.dhcp.start_ip = 10.0.0.10
Parameter- network.dhcp.end_ip
Description
Configuration File
<y0000000000xx>.cfg
Configures the end IP address that the IP phone assigns for the PC attached to the PC
268
Format
Default Value
Range
Example
Appendix port when the PC port is configured as Router.
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is not applicable to SIP-T19P and
SIP-T21P IP phones.
IP Address
10.0.0.100
Not Applicable network.dhcp.end_ip = 10.0.0.100
Replace Rule
Parameter- dialplan.replace.prefix.X
Description
Format
Default Value
Range
Example
Parameter- dialplan.replace.replace.X
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the string you want to replace.
X ranges from 1 to 100.
String
Blank
String within 32 characters dialplan.replace.prefix.1 = 123
Configuration File
<y0000000000xx>.cfg
Configures the alternate string instead of what the user enters.
X ranges from 1 to 100.
String
Blank
String within 32 characters dialplan.replace.replace.1 = 1
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Parameter- dialplan.replace.line_id.X
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Configures the desired line to apply this replace rule. The digit 0 stands for all lines.
X ranges from 1 to 100.
Note: Multiple line IDs are separated by commas. It is not applicable to SIP-T19P IP phones.
Integer
Blank (for all lines)
Valid values are:
0 to 6 (for SIP-T28P)
0 to 3 (for SIP-T26P/T22P)
0 to 2 (for SIP-T21P/T20P) dialplan.replace.line_id.1 = 1,2 Example
Dial-now
Parameter- dialplan.dialnow.rule.X
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the string used to match the numbers entered by the user.
When entered numbers match the predefined dial-now rule, the IP phone will automatically dial out the numbers without pressing the send key.
X ranges from 1 to 100.
String
Blank
String within 511 characters dialplan.dialnow.rule.1 = 123
Parameter- dialplan.dialnow.line_id.X
Description
Configuration File
<y0000000000xx>.cfg
Configures the desired line to apply this dial-now rule. The digit 0 stands for all lines.
X ranges from 1 to 100.
Note: Multiple line IDs are separated by
270
Format
Default Value
Range
Appendix commas. It is not applicable to SIP-T19P IP phones.
Integer
Blank (for all lines)
Valid values are:
0 to 6 (for SIP-T28P)
0 to 3 (for SIP-T26P/T22P)
0 to 2 (for SIP-T21P/T20P) dialplan.dialnow.line_id.1 = 1,2 Example
Parameter- phone_setting.dialnow_delay
Description
Format
Default Value
Range
Example
Area Code
Parameter- dialplan.area_code.code
Configuration File
<y0000000000xx>.cfg
Configures the delay time (in seconds) for the dial-now rule.
When entered numbers match the predefined dial-now rule, the IP phone will automatically dial out the entered number after the specified delay time.
Integer
1
1 to 14 phone_setting.dialnow_delay = 1
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the area code to add before the entered numbers.
String
Blank
String within 16 characters dialplan.area_code.code = 010
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Parameter- dialplan.area_code.min_len
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the minimum length of the entered numbers.
Integer
1
1 to 15 dialplan.area_code.min_len = 1
272
Parameter- dialplan.area_code.max_len
Description
Format
Default Value
Range
Example
Parameter- dialplan.area_code.line_id
Configuration File
<y0000000000xx>.cfg
Configures the maximum length of the entered numbers.
Note: The value must be larger than the minimum length.
Integer
15
1 to 15 dialplan.area_code.max_len = 15
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the desired line to apply this area code rule. The digit 0 stands for all lines.
Note: Multiple line IDs are separated by commas. It is not applicable to SIP-T19P IP phones.
Integer
Blank (for all lines)
Valid values are:
0 to 6 (for SIP-T28P)
0 to 3 (for SIP-T26P/T22P)
0 to 2 (for SIP-T21P/T20P) dialplan.area_code.line_id = 1,2
Appendix
Block Out
Parameter- dialplan.block_out.number.X
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the block out numbers.
X ranges from 1 to 10.
String
Blank
String within 32 characters dialplan.block_out.number.1 = 1234
Parameter- dialplan.block_out.line_id.X
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the desired line to apply this block out rule. The digit 0 stands for all lines.
X ranges from 1 to 10.
Note: Multiple line IDs are separated by commas. It is not applicable to SIP-T19P IP phones.
Integer
Blank (for all lines)
Valid values are:
0 to 6 (for SIP-T28P)
0 to 3 (for SIP-T26P/T22P)
0 to 2 (for SIP-T21P/T20P) dialplan.block_out.line_id.1 = 1,2,3
Parameter- phone_setting.common_power
_led_enable
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Enables or disables the power indicator LED to be turned on.
Boolean
1
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Range
Example
Valid values are:
0-Disabled (power indicator LED is off)
1-Enabled (power indicator LED is solid green) phone_setting.common_power_led_enable =
1
Parameter- phone_setting.ring_power_led_
flash_enable
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Enables or disables the power indicator LED to flash when the phone receives an incoming call.
If it is set to 0, the status of the power indicator
LED is determined by the value of the parameter
“phone_setting.common_power_led_enable”.
Boolean
1
Valid values are:
0-Disabled (power indicator LED does not flash)
1-Enabled (power indicator LED fast flashes
(300ms) green) phone_setting.ring_power_led_flash_enable =
1
Parameter- phone_setting.mail_power_led_
flash_enable
Configuration File
<y0000000000xx>.cfg
Description
Format
Enables or disables the power indicator LED to flash when the phone receives a voice mail or a text message.
If it is set to 0, the status of the power indicator
LED is determined by the value of the parameter
“phone_setting.common_power_led_enable”.
Boolean
274
Default Value
Range
Example
Appendix
0
Valid values are:
0-Disabled (power indicator LED does not flash)
1-Enabled (power indicator LED slow flashes
(1000ms) green) phone_setting.mail_power_led_flash_enable
= 0
Parameter- phone_setting.mute_power_led
_flash_enable
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Enables or disables the power indicator LED to flash when a call is mute.
If it is set to 0, the status of the power indicator
LED is determined by the value of the parameter
“phone_setting.common_power_led_enable”.
Boolean
1
Valid values are:
0-Disabled (power indicator LED does not flash)
1-Enabled (power indicator LED fast flashes
(300ms) green) phone_setting.mute_power_led_flash_enable
= 1
Parameter- phone_setting.hold_and_held_
power_led_flash_enable
Configuration File
<y0000000000xx>.cfg
Description
Enables or disables the power indicator LED to flash when a call is placed on hold or is held.
If it is set to 0, the status of the power indicator
LED is determined by the value of the parameter
“phone_setting.common_power_led_enable”.
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Format
Default Value
Range
Example
Boolean
0
Valid values are:
0-Disabled (power indicator LED does not flash)
1-Enabled (power indicator LED fast flashes
(500ms) green) phone_setting.hold_and_held_power_led_flas
h_enable = 0
Parameter- phone_setting.talk_and_dial_p
ower_led_enable
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Enables or disables the power indicator LED to be turned on when the phone is busy.
If it is set to 0, the status of the power indicator
LED is determined by the value of the parameter
“phone_setting.common_power_led_enable”.
Boolean
1
Valid values are:
0-Disabled (power indicator LED is off)
1-Enabled ( power indicator LED is solid green) phone_setting.talk_and_dial_power_led_enab
le = 1
276
Parameter- phone_setting.contrast
Description
Configuration File
<y0000000000xx>.cfg
Configures the contrast of the LCD screen.
For SIP-T28P IP phones, it configures the LCD’s contrast of the IP phone and connected EXP39.
For SIP-T26P IP phones, it configures the LCD’s contrast of the connected EXP39 only.
Format
Default Value
Range
Example
Appendix
For SIP-T19P and SIP-T21P IP phones, it configures the LCD’s contrast of the IP phone only.
Note: We recommend that you set the contrast of the LCD screen to 6 as a more comfortable level. It is only applicable to SIP-T19P, SIP-T21P and SIP-T28P IP phones, and EXP39 connected to SIP-T26P and SIP-T28P IP phones.
Integer
6
1 to 10 phone_setting.contrast = 6
Parameter- phone_setting.active_backlight
_level
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures the backlight idle intensity used to adjust the backlight intensity of the LCD screen
Level 3 is the brightest.
Note: It is only applicable to SIP-T28P IP phones and the connected EXP39.
Integer
2
1 to 3 phone_setting.active_backlight_level = 2
Parameter- phone_setting.backlight_time
Description
Configuration File
<y0000000000xx>.cfg
Configures the delay time to turn off the backlight when the IP phone is inactive.
If it is set to 60 (60s), the LCD backlight is turned off when the IP phone is inactive for 60 seconds.
Note: It is not applicable to SIP-T19P, SIP-T20P
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Format
Default Value
Range
Example and SIP-T21P IP phones.
Integer
30
Valid values are:
0-Always off
1-Always on
15-15s
30-30s
60-60s
120-120s
300-300s
600-600s
1800-1800s phone_setting.backlight_time = 30
Parameter- security.user_password
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the password of the user for web server access.
The IP phone uses “user” as the default user password.
Note: IP phones support ASCII characters
32-126(0x20-0x7E) only in passwords. username:new password user
String within 32 characters security.user_password = user:password123
278
Parameter- security.user_password
Description
Configuration File
<y0000000000xx>.cfg
Configures the password of the administrator for web server access.
The IP phone uses “admin” as the default
Format
Default Value
Range
Example
Appendix administrator password.
Note: IP phones support ASCII characters
32-126(0x20-0x7E) only in passwords. administrator username:new password admin
String within 32 characters security.user_password = admin:password000
Parameter- phone_setting.lock
Description
Configuration File
<y0000000000xx>.cfg
Configures the type of phone lock.
Menu Key: The Menu soft key and MESSAGE key are locked (For SIP-T20P, the MENU key is locked).
Function Keys: MESSAGE, RD, CONF, HOLD,
MUTE, TRAN, OK, X, navigation keys, soft keys, line keys and memory keys are locked (For
SIP-T22P/T21P, CONF, HOLD, MUTE and memory keys do not exist; For SIP-T20P, the
MUTE key, soft keys and memory keys do not exist, but the additional MENU and Directory keys are locked; For SIP-T19P, CONF, HOLD, OK,
X, memory keys and line keys do not exist, but the additional √ key is locked).
All Keys: All keys are locked except the volume key. You are only allowed to dial emergency numbers, reject incoming calls by pressing the
X key, answer incoming calls by lifting the handset, pressing the Speakerphone key, the
HEADSET key or the OK key, place an active call on hold by pressing the Hold soft key or the HOLD key, resume the held call by pressing the Resume soft key or the HOLD key, and end the call by hanging up the handset, pressing the Speakerphone key or pressing the X key (For SIP-T22P/T21P, HOLD key does not exist; For SIP-T20P, soft keys do not exist. For
SIP-T19P, HOLD and X keys do not exist).
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Format
Default Value
Range
If it is set to 0 (Disabled), IP phone lock feature is disabled.
Integer
0
Valid values are:
0-Disabled
1-Menu Key
2-Function Keys
3-All Keys phone_setting.lock = 1 Example
Parameter- phone_setting.phone_lock.unlo
ck_pin
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures a new unlock PIN. Once the IP phone is locked, you can use the default password “123” to unlock it. numeric characters
123 characters within 15 digits phone_setting.phone_lock.unlock_pin = 123
Parameter- phone_setting.phone_lock.lock
_time_out
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures the IP phone to automatically lock the keypad after a delay time (in seconds).
If it is set to 0 (0s), the keypad will not be locked automatically. In this case, you need to long press the pound key to lock the keypad.
Note: This parameter works only if the IP phone lock type is preset.
Integer
0
0 to 3600 phone_setting.phone_lock.lock_time_out = 8
280
Appendix
Parameter- local_time.manual_time_enabl
e
Configuration File
<MAC>.cfg
Description
Format
Default Value
Range
Example
Configures the phone to obtain time from NTP server or manual settings.
Integer
1
Valid values are:
0-Manual
1-NTP local_time.manual_time_enable = 1
NTP Server
Parameter- Configuration File local_time.manual_ntp_srv_prior <MAC>.cfg
Description
Enables or disables the phone to use manually configured NTP server preferentially.
Format
Default Value
Range
Example
Boolean
0
Valid values are:
0-Disabled (use the NTP server obtained by
DHCP preferentially)
1-Enabled local_time.manual_ntp_srv_prior = 0
Parameter- local_time.ntp_server1
Description
Format
Default Value
Range
Configuration File
<MAC>.cfg
Configures the IP address or the domain name of the primary NTP server.
IP Address or Domain Name cn.pool.ntp.org
String within 99 characters
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Example local_time.ntp_server1 = cn.pool.ntp.org
Parameter- local_time.ntp_server2
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the IP address or the domain name of the secondary NTP server. If the primary NTP server is not configured or cannot be accessed, the IP phone will request the time and date from the secondary NTP server.
IP Address or Domain Name cn.pool.ntp.org
String within 99 characters local_time.ntp_server2 = cn.pool.ntp.org
Parameter- local_time.interval
Description
Format
Default Value
Range
Example
Time Zone
Parameter- local_time.time_zone
Configuration File
<MAC>.cfg
Configures the IP phone to update time and date from the NTP server at regular intervals
(in seconds).
Integer
1000
15 to 86400 local_time.interval = 1000
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the time zone.
For more available time zones, refer to
Appendix B: Time Zones on page 257 .
String
+8
-11 to +13 local_time.time_zone = +8
282
Parameter- local_time.time_zone_name
Description
Format
Default Value
Range
Example
DST
Parameter- local_time.summer_time
Description
Format
Default Value
Range
Appendix
Configuration File
<MAC>.cfg
Configures the desired time zone name.
For more available time zone names, refer to
Appendix B: Time Zones on page 257 .
String
China(Beijing)
String within 32 characters local_time.time_zone_name = China(Beijing)
Configuration File
<MAC>.cfg
Enables or disables Daylight Saving Time
(DST) feature.
Integer
2
Valid values are:
0-Disabled
1-Enabled
2-Automatic local_time.summer_time = 2 Example
Parameter- local_time.dst_time_type
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the DST type.
Note: It works only if the parameter
“local_time.summer_time” is set to 1
(Enabled).
Integer
0
Valid values are:
0-By Date
1-By Week local_time.dst_time_type = 0
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Parameter- local_time.start_time
Description
Format
Default Value
Range
Example
Parameter- local_time.end_time
Configuration File
<MAC>.cfg
Configures the time to start DST.
If “local_time.dst_time_type” is set to 0 (By
Date), use the mapping:
MM: 1=Jan, 2=Feb,…, 12=Dec
DD:1=the first day in a month,…, 31= the last day in a month
HH:0=1am, 1=2am,…, 23=12pm
If “local_time.dst_time_type” is set to 1 (By
Week), use the mapping:
Month: 1=Jan, 2=Feb,…, 12=Dec
Week of Month: 1=the first week in a month,…,
5=the last week in a month
Day of Week: 1=Mon, 2=Tues,…, 7=Sun
Hour of Day: 0=1am, 1=2am,…, 23=12pm
Note: It works only if the parameter
“local_time.summer_time” is set to 1
(Enabled).
The value formats are:
MM/DD/HH (For By Date)
Month/Week of Month/Day of Week/Hour of Day (For By Week)
1/1/0
1to 12/1 to 31/0 to 23 (for By Date)
1 to 12/1 to 5/1 to 7/0 to 23 (for By Week) local_time.start_time = 1/1/0
Description
Configuration File
<MAC>.cfg
Configures the time to end DST.
If “local_time.dst_time_type” is set to 0 (By
Date), use the mapping:
MM: 1=Jan, 2=Feb,…, 12=Dec
DD:1=the first day in a month,…, 31= the last day in a month
HH:0=1am, 1=2am,…, 23=12pm
284
Format
Default Value
Range
Example
Parameter- local_time.dhcp_time
Description
Format
Default Value
Range
Example
Appendix
If “local_time.dst_time_type” is set to 1 (By
Week), use the mapping:
Month: 1=Jan, 2=Feb,…, 12=Dec
Week of Month: 1=the first week in a month,…,
5=the last week in a month
Day of Week: 1=Mon, 2=Tues,…, 7=Sun
Hour of Day: 0=1am, 1=2am,…, 23=12pm
Note: It works only if the parameter
“local_time.summer_time” is set to 1
(Enabled).
The value formats are:
MM/DD/HH (For By Date)
Month/Week of Month/Day of Week/Hour of Day (For By Week)
12/31/23
1to 12/1 to 31/0 to 23 (For By Date)
1 to 12/1 to 5/1 to 7/0 to 23 (For By Week) local_time.end_time = 12/31/23
Configuration File
<MAC>.cfg
Enables or disables the phone to update time with the offset time obtained from the DHCP server.
Note: It is only available to offset from GMT 0.
Boolean
0
Valid values are:
0-Disabled
1-Enabled local_time.dhcp_time = 0
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Parameter- local_time.offset_time
Description
Format
Default Value
Range
Example
Time Format
Parameter- local_time.time_format
Configuration File
<MAC>.cfg
Configures the offset time (in minutes).
Integer
Blank
-300 to +300 local_time.offset_time = 120
Description
Format
Default Value
Range
Configuration File
<MAC>.cfg
Configures the time format.
If it is set to 0 (12 Hour), the time display will use 12 hour format.
If it is set to 1 (24 Hour), the time display will use 24 hour format.
Integer
1
Valid values are:
0-12 Hour
1-24 Hour local_time.time_format = 1 Example
Date Format
Parameter- local_time.date_format
Description
Format
Default Value
Range
Configuration File
<MAC>.cfg
Configures the date format.
IP phones support various date formats. You can change the desired format according to your requirement.
Integer
0
Note: For SIP-T20P IP phones, the default value is 7.
For SIP-T28P/T26P/T21P/T22P/T19P IP phones:
286
Example
Parameter- gui_lang.url
Description
Format
Default Value
Range
Example
Appendix
Valid values are:
0-WWW MMM DD
1-DD-MMM-YY
2-YYYY-MM-DD
3-DD/MM/YYYY
4-MM/DD/YY
5-DD MMM YYYY
6-WWW DD MMM
For SIP-T20P IP phones:
7-MM DD YY
8-DD MM YY
9-YY MM DD
Note: “WWW” represents the abbreviation of the week, “DD” represents a two-digit day,
“MMM” represents the first three letters of the month, “YYYY” represents a four-digit year, and “YY” represents a two-digit year which is not displayed on the LCD screen of SIP-T20P IP phones. local_time.date_format = 0
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the language pack.
Note: The language packs you load are dependent on available language packs from the provisioning server. You can download the language pack to the phone user interface only.
URL
Blank
String within 511 characters
The following example uses HTTP to download the language pack
“lang+English.txt” from the provisioning server 192.168.10.25. gui_lang.url =
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Parameter- lang.gui
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Configures the language used on the phone user interface.
String
English
Valid values are:
English
Chinese_S (only applicable to SIP-T19P and
SIP-T21P IP phones)
Chinese_T (only applicable to SIP-T19P and
SIP-T21P IP phones)
German
French
Italian
Portuguese
Polish
Spanish
Turkish lang.gui = English Example
Parameter- lang.wui
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Configures the language used on the web user interface.
Note: The default language used on the web user interface depends on the language preferences of your browser. If the language of your browser is not supported by the IP phone, the web user interface will use English by default.
String
Blank
Valid values are:
288
Example
Appendix
English
Chinese_S (only applicable to SIP-T19P and
SIP-T21P IP phones)
German
French (not applicable to SIP-T19P and SIP-T21P
IP phones)
Italian
Portuguese (not applicable to SIP-T19P and
SIP-T21P IP phones)
Spanish (not applicable to SIP-T19P and
SIP-T21P IP phones)
Turkish lang.wui = English
Parameter- phone_setting.lcd_logo.mode
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Configures the logo mode of the LCD screen.
If it is set to 0 (Disabled), the IP phone is not allowed to display a logo.
If it is set to 1 (System logo), the LCD screen will display the system logo.
If it is set to 2 (Custom logo), the LCD screen will display the custom logo (you need to upload a custom logo file to the phone).
For SIP-T20P IP phones:
Enables or disables a text logo.
If it is set to 0 (Disabled), the IP phone is not allowed to display a text logo.
If it is set to 1 (Enabled), the LCD screen will display the custom text logo.
Integer
0
Note: For SIP-T28 IP phones, the default value is
1.
Valid values are:
0-Disabled
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Example
1-System logo
2-Custom logo
Note: For SIP-T28 IP phones, valid values are
1(System logo) and 2(Custom logo). For
SIP-T20P IP phones, valid values are
0(Disabled) and 1(Enabled). phone_setting.lcd_logo.mode = 1
Parameter- lcd_logo.url
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the access URL of custom logo file.
Note: It is not applicable to SIP-T20P IP phones.
URL
Blank
String within 511 characters
The following example uses HTTP to download the custom logo file (logo.dob) from the provisioning server 192.168.10.25. lcd_logo.url = http://192.168.10.25/logo.dob
Parameter- phone_setting.lcd_logo.text
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures a text logo.
Note: It is only applicable to SIP-T20P IP phones.
String
Yealink
String within 15 characters phone_setting.lcd_logo.text = Yealink
290
Parameter- features.key_as_send
Description
Configuration File
<y0000000000xx>.cfg
Configures the "#" or "*" key as the send key.
Format
Default Value
Range
Example
Parameter- features.key_tone
Description
Format
Default Value
Range
Example
Parameter- features.send_key_tone
Description
Format
Default Value
Appendix
If it is set to 0 (Disabled), neither “#” nor “*” can be used as a send key.
If it is set to 1 (# key), the pound key is used as the send key.
If it is set to 2 (* key), the asterisk key is used as the send key.
Integer
1
Valid values are:
0-Disabled
1-# key
2-* key features.key_as_send = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to play a tone when a user presses a key.
If it is set to 1 (Enabled), the IP phone will play a tone when a user presses a key.
Boolean
1
0-Disabled
1-Enabled features.key_tone = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to play a tone when a user presses a send key.
If it is set to 1 (Enabled), the IP phone will play a tone when a user presses a send key.
Note: It works only if the parameter
“features.key_tone” is set to 1 (Enabled).
Boolean
1
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Range
Example
Valid values are:
0-Disabled
1-Enabled features.send_key_tone = 1
Parameter- features.hotline_number
Description
Format
Default Value
Range
Example
Parameter- features.hotline_delay
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Configures the hotline number.
It configures a number that the IP phone automatically dials out when lifting the handset, pressing the speakerphone key or the line key. Leaving it blank disables hotline feature.
String
Blank
String within 32 characters features.hotline_number = 3601
Configuration File
<y0000000000xx>.cfg
Configures the waiting time (in seconds) the IP phone automatically dials out the hotline number.
If it is set to 0 (0s), the IP phone will immediately dial out the preconfigured hotline number when you lift the handset, press the speakerphone key or press the line key.
If it is set to a value greater than 0, the IP phone will wait the specified seconds before dialing out the predefined hotline number when you lift the handset, press the speakerphone key or press the line key.
Integer
4
0 to 10
292
Example
Appendix features.hotline_delay = 4
Parameter- features.save_call_history
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to save call log.
If it is set to 0 (Disabled), the IP phone cannot log the placed calls, received calls, missed calls and the forwarded calls in the call log lists.
Boolean
1
Valid values are:
0-Disabled
1-Enabled features.save_call_history = 1
Parameter- account.X.missed_calllog
Description
Format
Default Value
Range
Configuration File
<MAC>.cfg
Enables or disables missed call log feature for account X.
If it is set to 0 (Disabled), there is no indicator displaying on the LCD screen, the IP phone does not log the missed call in the Missed
Calls list.
If it is set to 1 (Enabled), a prompt message
"<number> New Missed Call(s)" along with an indicator icon is displayed on the IP phone idle screen when the IP phone misses calls.
X ranges from 1 to 6.
Boolean
1
Valid values are:
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Example
0-Disabled
1-Enabled account.1.missed_calllog = 1
Parameter- phone_setting.predial_autodial
Configuration File
<y0000000000xx>.cfg
Description
Format
Enables or disables live dialpad feature.
If it is set to 1 (Enabled), the IP phone will automatically dial out the entered phone number without having to press any key.
Boolean
Default Value
Range
Example
0
Valid values are:
0-Disabled
1-Enabled phone_setting.predial_autodial = 1
Parameter- phone_setting.inter_digit_time
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the time (in seconds) for the phone to automatically dial out the entered digits without pressing any other key.
Note: It works only if the parameter
“phone_setting.predial_autodial” is set to 1
(Enabled).
Integer
4
1 to 14 phone_setting.inter_digit_time = 4
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Appendix
Parameter- call_waiting.enable
Description
Format
Default Value
Range
Example
Parameter- call_waiting.tone
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables call waiting feature.
If it is set to 0 (Disabled), a new incoming call is automatically rejected by the IP phone with a busy message while during a call.
If it is set to 1 (Enabled), the LCD screen will present a new incoming call while during a call.
Boolean
1
Valid values are:
0-Disabled
1-Enabled call_waiting.enable = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables the playing of a call waiting tone when the IP phone receives an incoming call during a call.
If it is set to 1 (Enabled), the IP phone will perform an audible indicator when receiving a new incoming call during a call.
Note: It works only if the parameter
“call_waiting.enable” is set to 1 (Enabled).
Boolean
1
Valid values are:
0-Disabled
1-Enabled call_waiting.tone = 1
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Parameter- call_waiting.on_code
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the call waiting on code to activate the server-side call waiting feature.
String
Blank
String within 32 characters call_waiting.on_code = *72
Parameter- call_waiting.off_code
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the call waiting off code to deactivate the server-side call waiting feature.
String
Blank
String within 32 characters call_waiting.off_code = *73
Parameter- auto_redial.enable
Description
296
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to automatically redial the called number when it is busy.
If it is set to 1 (Enabled), the IP phone will dial the previous dialed out number automatically when the dialed number is busy.
Boolean
0
Valid values are:
0-Disabled
1-Enabled auto_redial.enable = 1
Parameter- auto_redial.interval
Description
Format
Default Value
Range
Example
Parameter- auto_redial.times
Description
Format
Default Value
Range
Example
Appendix
Configuration File
<y0000000000xx>.cfg
Configures the interval (in seconds) for the IP phone to wait between redials.
The IP phone redials the dialed number at regular intervals till the callee answers the call.
Integer
10
1 to 300 auto_redial.interval = 10
Configuration File
<y0000000000xx>.cfg
Configures the redial times for the IP phone.
The IP phone tries to redial the dialed number as many times as configured till the callee answers the call.
Integer
10
1 to 300 auto_redial.times = 10
Parameter- account.X.auto_answer
Description
Configuration File
<MAC>.cfg
Enables or disables auto answer feature for account X.
If it is set to 1 (Enabled), the IP phone can automatically answer an incoming call.
X ranges from 1 to 6.
Note: The IP phone cannot automatically answer the incoming call during a call even if auto answer is enabled.
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Format
Default Value
Range
Example
Parameter- features.auto_answer_delay
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.auto_answer = 1
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the delay time (in seconds) before the phone automatically answers an incoming call.
Integer
1
1 to 4 features.auto_answer_delay = 1
298
Parameter- features.call_completion_enable
Configuration File
<y0000000000xx>.cfg
Description
Enables or disables call completion feature.
If a user places a call and the callee is temporarily not available to answer the call, call completion feature allows notifying the user when the callee becomes available to receive a call.
If it is set to 1 (Enabled), the caller is notified when the callee becomes available to receive a call.
Format
Default Value
Range
Boolean
0
Valid values are:
0-Disabled
1-Enabled
Example
Appendix features.call_completion_enable = 1
Parameter- account.X.anonymous_call
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Enables or disables anonymous call feature for account X.
If it is set to 1 (Enabled), the IP phone will block its identity from showing up to the callee when placing a call. The callee’s phone LCD screen presents anonymous instead of the caller’s identity.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.anonymous_call = 1
Parameter- account.X.send_anonymous_co
de
Configuration File
<MAC>.cfg
Description
Format
Default Value
Configures the phone to send anonymous on/off code to activate/deactivate the server-side anonymous call feature for account X.
If it is set to 0 (Off Code), the IP phone will send anonymous off code to deactivate the server-side anonymous call feature.
If it is set to 1 (On Code), the IP phone will send anonymous on code to activate the server-side anonymous call feature.
X ranges from 1 to 6.
Boolean
0
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Range
Example
Valid values are:
0-Off Code
1-On Code account.1.send_anonymous_code = 0
Parameter- account.X.anonymous_call_onc
ode
Configuration File
<MAC>.cfg
Description
Configures the anonymous call on code to activate the server-side anonymous call feature for account X.
X ranges from 1 to 6.
Note: It works only if the parameter
“account.X.send_anonymous_code” is set to 1
(Enabled).
Format
Default Value
Range
Example
String
Blank
String within 32 characters account.1.anonymous_call_oncode = *72
Parameter- account.X.anonymous_call_off
code
Configuration File
<MAC>.cfg
Description
Format
Default Value
Range
Example
Configures the anonymous call off code to deactivate the server-side anonymous call feature for account X.
X ranges from 1 to 6.
Note: It works only if the parameter
“account.X.send_anonymous_code” is set to 1
(Enabled).
String
Blank
String within 32 characters account.1.anonymous_call_offcode = *73
300
Appendix
Parameter- account.X.reject_anonymous_c
all
Configuration File
<MAC>.cfg
Description
Format
Default Value
Range
Example
Enables or disables anonymous call rejection feature for account X.
If it is set to 1 (Enabled), the IP phone will automatically reject incoming calls from users enabled anonymous call feature. The anonymous user’s phone LCD screen presents
“Anonymity Disallowed”.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.reject_anonymous_call = 1
Parameter- account.X.anonymous_reject_o
ncode
Configuration File
<MAC>.cfg
Description
Format
Default Value
Range
Example
Configures the anonymous call rejection on code to activate the server-side anonymous call rejection feature for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.anonymous_reject_oncode = *74
Parameter- account.X.anonymous_reject_of
fcode
Description
Configuration File
<MAC>.cfg
Configures the anonymous call rejection off
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Format
Default Value
Range
Example code to deactivate the server-side anonymous call rejection feature for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.anonymous_reject_offcode = *75
302
Return Message When DND
Parameter- features.dnd_refuse_code
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Configures a return code and reason of SIP response messages when rejecting an incoming call by DND. A specific reason is displayed on the caller’s phone LCD screen.
If it is set to 486 (Busy here), the caller’s phone
LCD screen will display the reason “Busy here” when the callee enables DND feature.
Integer
480
Valid values are:
404-No Found
480-Temporarily not available
486-Busy here features.dnd_refuse_code = 480 Example
DND Mode
Parameter- features.dnd_mode
Description
Configuration File
<y0000000000xx>.cfg
Configures the DND mode for the IP phone.
If it is set to 0 (Phone), DND feature is effective for the IP phone.
If it is set to 1 (Custom), you can configure DND feature for each account.
Format
Default Value
Range
Example
DND in Phone Mode
Parameter- features.dnd.enable
Description
Format
Default Value
Range
Example
Parameter- features.dnd.on_code
Description
Format
Default Value
Range
Example
Parameter- features.dnd.off_code
Description
Format
Appendix
Integer
0
Valid values are:
0-Phone
1-Custom features.dnd_mode = 0
Configuration File
<y0000000000xx>.cfg
Enables or disables DND feature.
If it is set to 1 (Enabled), the IP phone will reject incoming calls on all accounts.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.dnd.enable = 1
Configuration File
<y0000000000xx>.cfg
Configures the DND on code to activate the server-side DND feature.
String
Blank
String within 32 characters features.dnd.on_code = *71
Configuration File
<y0000000000xx>.cfg
Configures the DND off code to deactivate the server-side DND feature.
String
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Default Value
Range
Example
DND in Custom Mode
Parameter- account.X.dnd.enable
Blank
String within 32 characters features.dnd.off_code = *72
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Enables or disables DND feature for account X.
If it is set to 1 (Enabled), the IP phone will reject incoming calls on account X.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.dnd.enable = 1
Parameter- account.X.dnd.on_code
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the DND on code to activate the server-side DND feature for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.dnd.on_code = *73
Parameter- account.X.dnd.off_code
Description
Format
Configuration File
<MAC>.cfg
Configures the DND off code to deactivate the server-side DND feature for account X.
X ranges from 1 to 6.
String
304
Default Value
Range
Example
Appendix
Blank
String within 32 characters account.1.dnd.off_code = *74
Parameter- features.busy_tone_delay
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures a period of time (in seconds) for which the busy tone is audible on the IP phone.
When one party releases the call, a busy tone is audible to the other party indicating that the call connection breaks.
If it is set to 3 (3s), a busy tone is audible for 3 seconds on the IP phone.
Integer
0
Valid values are:
0-0s
3-3s
5-5s features.busy_tone_delay = 0
Parameter- features.normal_refuse_code
Description
Format
Default Value
Configuration File
<y0000000000xx>.cfg
Configures a return code and reason of SIP response messages when rejecting an incoming call. A specific reason is displayed on the caller’s phone LCD screen.
If it is set to 486 (Busy here), the caller’s phone LCD screen will display the message
“Busy here” when the callee rejects the incoming call.
Integer
486
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Range
Example
Valid values are:
404-No Found
480-Temporarily not available
486-Busy here features.normal_refuse_code = 486
Parameter- phone_setting.is_deal180
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to deal with the 180 SIP message received after the 183
SIP message.
If it is set to 1 (Enabled), the IP phone will resume and play the local ringback tone upon a subsequent 180 message received.
Boolean
1
Valid values are:
0-Disabled
1-Enabled phone_setting.is_deal180 = 1
306
Parameter- sip.use_out_bound_in_dialog
Description
Format
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to send the
SIP requests to the outbound proxy server.
If it is set to 1 (Enabled), all the SIP request messages from the IP phone will be forced to send to the outbound proxy server.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
Default Value
Range
Example
Appendix
1
Valid values are:
0-Disabled
1-Enabled sip.use_out_bound_in_dialog = 1
Parameter- account.X.advanced.timer_t1
Description
Format
Default Value
Range
Example
Parameter- account.X.advanced.timer_t2
Configuration File
<MAC>.cfg
Configures the SIP session timer T1 (in seconds) for account X.
T1 is an estimate of the Round Trip Time (RTT) of transactions between a SIP client and SIP server.
X ranges from 1 to 6.
Float
0.5
0.5 to 10 account.1.advanced.timer_t1 = 0.5
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the session timer T2 (in seconds) for account X.
T2 represents the maximum retransmit interval for non-INVITE requests and INVITE responses.
X ranges from 1 to 6.
Float
4
2 to 40 account.1.advanced.timer_t2 = 4
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Parameter- account.X.advanced.timer_t4
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the session timer of T4 (in seconds) for account X.
T4 represents the maximum duration a message will remain in the network.
X ranges from 1 to 6.
Float
5
2.5 to 60 account.1.advanced.timer_t4 = 5
308
Parameter- account.X.session_timer.enable
Description
Format
Default Value
Range
Example
Parameter- account.X.session_timer.expires
Description
Configuration File
<MAC>.cfg
Enables or disables the session timer for account X.
If it is set to 1 (Enabled), IP phone will send periodic re-INVITE requests to refresh the session during a call.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.session_timer.enable = 1
Configuration File
<MAC>.cfg
Configures the IP phone to refresh the session during a call at regular intervals (in seconds) for account X.
If it is set to 1800 (1800s), the IP phone will
Format
Default Value
Range
Example
Integer
1800
30 to 7200
Appendix refresh the session during a call before 1800 seconds.
X ranges from 1 to 6. account.1.session_timer.expires = 1800
Parameter- account.X.session_timer.refresher
Configuration File
<MAC>.cfg
Description
Format
Default Value
Range
Configures the session timer refresher for account X.
If it is set to 0 (UAC), refreshing the session is performed by the IP phone.
If it is set to 1 (UAS), refreshing the session is performed by a SIP server.
X ranges from 1 to 6.
Integer
0
Valid values are:
0-UAC
1-UAS
Example account.1.session_timer.refresher = 0
Parameter- features.play_hold_tone.enable
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to play a tone when there is a hold call on the IP phone.
Boolean
1
Valid values are:
0-Disabled
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1-Enabled features.play_hold_tone.enable = 1 Example
Parameter- features.play_hold_tone.delay
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the interval (in seconds) at which the IP phone plays a hold tone.
If it is set to 30 (30s), the IP phone will play a hold tone every 30 seconds when there is a hold call on the IP phone.
Note: It works only if the parameter
“features.play_hold_tone.enable” is set to 1
(Enabled).
Integer
30
3 to 3600 features.play_hold_tone.delay = 30
Parameter- sip.rfc2543_hold
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures whether RFC 2543 (c=0.0.0.0) outgoing hold signaling is used.
If it is set to 0 (Disabled), SDP media direction attributes (such as a=sendonly) per RFC 3264 is used when placing a call on hold.
If it is set to 1 (Enabled), SDP media connection address c=0.0.0.0 per RFC 2543 is used when placing a call on hold.
Boolean
0
Valid values are:
0-Disabled
1-Enabled sip.rfc2543_hold = 0
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Appendix
Call Forward Mode
Parameter- features.fwd_mode
Description
Format
Default Value
Range
Example
Call Forward in Phone Mode
Always Forward
Parameter- forward.always.enable
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the call forward mode for the IP phone.
If it is set to 0 (Phone), call forward feature is effective for the IP phone.
If it is set to 1 (Custom), you can configure call forward feature for each account.
Integer
0
Valid values are:
0-Phone
1-Custom features.fwd_mode = 0
Configuration File
< y0000000000xx >.cfg
Enables or disables always forward feature.
If it is set to 1 (Enabled), incoming calls are forwarded to the destination number immediately.
Boolean
0
Valid values are:
0-Disabled
1-Enabled forward.always.enable = 1
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Parameter- forward.always.target
Description
Format
Default Value
Range
Example
Configuration File
< y0000000000xx >.cfg
Configures the destination number of the always forward.
String
Blank
String within 32 characters forward.always.target = 3601
Parameter- forward.always.on_code
Description
Format
Default Value
Range
Example
Configuration File
< y0000000000xx >.cfg
Configures the always forward on code to activate the server-side always forward feature.
String
Blank
String within 32 characters forward.always.on_code = *72
Parameter- forward.always.off_code
Description
Format
Default Value
Range
Example
Busy Forward
Parameter- forward.busy.enable
Description
Configuration File
< y0000000000xx >.cfg
Configures the always forward off code to deactivate the server-side always forward feature.
String
Blank
String within 32 characters forward.always.off_code = *73
Configuration File
< y0000000000xx >.cfg
Enables or disables busy forward feature.
If it is set to 1 (Enabled), incoming calls are
312
Format
Default Value
Range
Example
Parameter- forward.busy.target
Description
Format
Default Value
Range
Example
Parameter- forward.busy.on_code
Description
Format
Default Value
Range
Example
Parameter- forward.busy.off_code
Description
Format
Appendix forwarded to the destination number when the callee is busy.
Boolean
0
Valid values are:
0-Disabled
1-Enabled forward.busy.enable = 1
Configuration File
< y0000000000xx >.cfg
Configures the destination number of the busy forward.
String
Blank
String within 32 characters forward.busy.target = 3602
Configuration File
< y0000000000xx >.cfg
Configures the busy forward on code to activate the server-side busy forward feature.
String
Blank
String within 32 characters forward.busy.on_code = *74
Configuration File
< y0000000000xx >.cfg
Configures the busy forward off code to deactivate the server-side busy forward feature.
String
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Default Value
Range
Example
No Answer Forward
Parameter- forward.no_answer.enable
Blank
String within 32 characters forward.busy.off_code = *75
Description
Format
Default Value
Range
Example
Configuration File
< y0000000000xx >.cfg
Enables or disables no answer forward feature.
If it is set to 1 (Enabled), incoming calls are forward to the destination number after a period of ring time.
Boolean
0
Valid values are:
0-Disabled
1-Enabled forward.no_answer.enable = 1
Parameter- forward.no_answer.target
Description
Format
Default Value
Range
Example
Configuration File
< y0000000000xx >.cfg
Configures the destination number of the no answer forward.
String
Blank
String within 32 characters forward.no_answer.target = 3603
Parameter- forward.no_answer.timeout
Description
Format
Configuration File
< y0000000000xx >.cfg
Configures ring times (N) to wait before forwarding incoming calls.
Incoming calls will be forwarded when not answered after N*6 seconds.
Integer
314
Default Value
Range
Example
Parameter- forward.no_answer.on_code
Description
Format
Default Value
Range
Example
Appendix
2
0 to 20 forward.no_answer.timeout = 2
Configuration File
< y0000000000xx >.cfg
Configures the no answer forward on code to activate the server-side no answer forward feature.
String
Blank
String within 32 characters forward.no_answer.on_code = *76
Parameter- forward.no_answer.off_code
Description
Format
Default Value
Range
Example
Configuration File
< y0000000000xx >.cfg
Configures the no answer forward off code to deactivate the server-side no answer forward feature.
String
Blank
String within 32 characters forward.no_answer.off_code = *77
Call Forward in Custom Mode
Always Forward
Parameter- account.X.always_fwd.enable
Description
Configuration File
<MAC>.cfg
Enables or disables always forward feature for account X.
If it is set to 1 (Enabled), incoming calls to the account X are forwarded to the destination number immediately.
X ranges from 1 to 6.
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Format
Default Value
Range
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.always_fwd.enable = 1 Example
Parameter- account.X.always_fwd.target
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the destination number of the always forward for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.always_fwd.target = 3601
Parameter- account.X.always_fwd.on_code
Description
Format
Default Value
Range
Example
Parameter- account.X.always_fwd.off_code
Description
Configuration File
<MAC>.cfg
Configures the always forward on code to activate the server-side always forward feature for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.always_fwd.on_code = *72
Configuration File
<MAC>.cfg
Configures the always forward off code to deactivate the server-side always forward feature for account X.
X ranges from 1 to 6.
316
Format
Default Value
Range
Example
Busy Forward
Parameter- account.X.busy_fwd.enable
Appendix
String
Blank
String within 32 characters account.1.busy_fwd.off_code = *73
Description
Format
Default Value
Range
Configuration File
<MAC>.cfg
Enables or disables busy forward feature for account X.
If it is set to 1 (Enabled), incoming calls to the account X are forwarded to the destination number when the callee is busy.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.busy_fwd.enable = 1 Example
Parameter- account.X.busy_fwd.target
Description
Format
Default Value
Range
Example
Parameter- account.X.busy_fwd.on_code
Description
Configuration File
<MAC>.cfg
Configures the destination number of the busy forward for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.busy_fwd.target = 3602
Configuration File
<MAC>.cfg
Configures the busy forward on code to
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Format
Default Value
Range
Example activate the server-side busy forward feature for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.busy_fwd.on_code = *74
Parameter- account.X.busy_fwd.off_code
Description
Format
Default Value
Range
Example
No Answer Forward
Parameter- account.X.timeout_fwd.enable
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the busy forward off code to deactivate the server-side busy forward feature for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.busy_fwd.off_code = *75
Configuration File
<MAC>.cfg
Enables or disables no answer forward feature for account X.
If it is set to 1 (Enabled), incoming calls to the account X are forward to the destination number after a period of ring time.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.timeout_fwd.enable = 1
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Appendix
Parameter- account.X.timeout_fwd.target
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the destination number of the no answer forward for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.timeout_fwd.target = 3603
Parameter- account.X.timeout_fwd.timeout
Description
Format
Default Value
Range
Example
Parameter- account.X.timeout_fwd.on_code
Configuration File
<MAC>.cfg
Configures ring times (N) to wait before forwarding incoming calls for account X.
Incoming calls will be forwarded when not answered after N*6 seconds
X ranges from 1 to 6.
Integer
2
0 to 20 account.1.timeout_fwd.timeout = 2
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the no answer forward on code to activate the server-side no answer forward feature for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.timeout_fwd.on_code = *76
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Parameter- account.X.timeout_fwd.off_code
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the no answer forward off code to activate the server-side no answer forward feature for account X.
X ranges from 1 to 6.
String
Blank
String within 32 characters account.1.timeout_fwd.off_code = *77
Fwd International
Parameter- forward.international.enable
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to forward an incoming call to an international phone number.
Boolean
1
Valid values are:
0-Disabled
1-Enabled forward.international.enable = 1
320
Parameter- transfer.blind_tran_on_hook_ena
ble
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Enables or disables the IP phone to complete the blind transfer through on-hook.
Boolean
1
Valid values are:
0-Disabled
Appendix
Example
Parameter- transfer.on_hook_trans_enable
Description
Format
Default Value
Range
Example
1-Enabled transfer.blind_tran_on_hook_enable = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to complete the semi-attended transfer or the attended transfer through on-hook.
Boolean
1
Valid values are:
0-Disabled
1-Enabled transfer.on_hook_trans_enable = 1
Parameter- transfer.semi_attend_tran_enable
Configuration File
<y0000000000xx>.cfg
Description
Configures whether to display the missed call prompt on the destination party’s phone when performing a semi-attended transfer.
Format
Default Value
Range
Example
Boolean
1
Valid values are:
0-Enabled
1-Disabled transfer.semi_attend_tran_enable = 1
Parameter- account.X.conf_type
Description
Configuration File
<MAC>.cfg
Configures the conference type for account
X.
If it is set to 0 (Local Conference), conferences are set up on the IP phone
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Format
Default Value
Range
Example locally.
If it is set to 2 (Network Conference), conferences are set up by the server.
X ranges from 1 to 6.
Integer
0
Valid values are:
0-Local Conference
2-Network Conference account.1.conf_type = 0
Parameter- account.X.conf_uri
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the conference URI for account X.
X ranges from 1 to 6.
Note: It works only if the parameter
“account.X.conf_type” is set to 2 (Network
Conference).
SIP URI
Blank
SIP URI within 511 characters account.1.conf_uri = [email protected]
322
Parameter- transfer.tran_others_after_conf_e
nable
Configuration File
<y0000000000xx>.cfg
Description
Enables or disables Transfer on Conference
Hang Up feature.
If enabled, the other two parties remain connected when the conference initiator drops the conference call.
Note: It is only applicable to the local conference.
Format
Default Value
Range
Example
Appendix
Boolean
0
Valid values are:
0-Disabled
1-Enabled transfer.tran_others_after_conf_enable = 1
Phone Basis
Parameter- features.pickup.direct_pickup_e
nable
Configuration File
<MAC>.cfg
Description
Format
Default Value
Range
Example
Enables or disables the IP phone to display the DPickup soft key when the IP phone is off-hook.
Note: It is not applicable to SIP-T20P IP phones.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.pickup.direct_pickup_enable = 1
Parameter- features.pickup.direct_pickup_co
de
Configuration File
<MAC>.cfg
Description
Format
Default Value
Configures the directed call pickup code on a phone basis.
Note: The directed call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
String
Blank
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Range
Example
Per-line Basis
Parameter- account.X.direct_pickup_code
String within 32 characters features.pickup.direct_pickup_code = *97
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the directed call pickup code on a per-line basis.
X ranges from 1 to 6.
Note: The directed call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
String
Blank
String within 32 characters account.1.direct_pickup_code = *68
324
Phone Basis
Parameter- features.pickup.group_pickup_en
able
Configuration File
<MAC>.cfg
Description
Enables or disables the IP phone to display the GPickup soft key when the IP phone is off-hook.
Note: It is not applicable to SIP-T20P IP phones.
Format
Default Value
Range
Example
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.pickup.group_pickup_enable = 1
Appendix
Parameter- features.pickup.group_pickup_co
de
Configuration File
<MAC>.cfg
Description
Format
Default Value
Range
Example
Configures the group call pickup code on a phone basis.
Note: The group call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
String
Blank
String within 32 characters features.pickup.group_pickup_code = *98
Per-line Basis
Parameter- account.X.group_pickup_code
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the group call pickup code on a per-line basis.
X ranges from 1 to 6.
Note: The group call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
String
Blank
String within 32 characters account.1.group_pickup_code = *69
Parameter- account.X.dialoginfo_callpickup
Description
Configuration File
<MAC>.cfg
Configures Dialog Info Call Pickup feature for account X.
If it is set to 1 (Enabled), call pickup is implemented through SIP signals.
X ranges from 1 to 6.
325
Parameter- wui.http_enable
Description
Format
Default Value
Range
Example
Parameter- network.port.http
Description
326
Format
Default Value
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Format
Default Value
Range
Example
Note: It is not applicable to SIP-T19P IP phones.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.dialoginfo_callpickup = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to access its web user interface using HTTP protocol.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
1
Valid values are:
0-Disabled
1-Enabled wui.http_enable = 1
Configuration File
<y0000000000xx>.cfg
Configures the HTTP port used to access the web user interface of the IP phone.
The default HTTP port is 80.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
80
Range
Example
Parameter- wui.https_enable
Description
Format
Default Value
Range
Example
Parameter- network.port.https
Description
Format
Default Value
Range
Example
Appendix
1 to 65535 network.port.http = 80
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to access its web user interface using HTTPS protocol.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
1
Valid values are:
0-Disabled
1-Enabled wui.https_enable = 1
Configuration File
<y0000000000xx>.cfg
Configures the HTTPS port used to access the web user interface of the IP phone.
The default HTTPS port is 443.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
443
1 to 65535 network.port.https = 443
Parameter- account.X.cid_source
Description
Configuration File
<MAC>.cfg
Configures the presentation of the caller
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Format
Default Value
Range
Example identity for account X.
0-FROM (Derives the name and number of the caller from the “From” header).
1-PAI (Derives the name and number of the caller from the “PAI” header. If the server does not send the “PAI” header, displays
“anonymity” on the callee’s phone).
2-PAI-FROM (Derives the name and number of the caller from the “PAI” header preferentially. If the server does not send the
“PAI” header, derives from the “From” header).
3-RPID-PAI-FROM
4-PAI-RPID-FROM
5-RPID-FROM
X ranges from 1 to 6.
Integer
0
0 to 5 account.1.cid_source = 0
328
Parameter- account.X.cp_source
Description
Configuration File
<MAC>.cfg
Configures the presentation of the callee’s identity for account X.
0-PAI-RPID (Derives the name and number of the callee from the “PAI” header preferentially. If the server does not send the
“PAI” header, derives from the “RPID” header).
1-Dialed Digits (Preferentially displays the dialed digits on the caller’s phone).
2-RFC 4916 (Derives the name and number of the callee from “From” header in the
Update message).
When the RFC 4916 is enabled on the IP phone, the caller sends the SIP request message which contains the from-change
Format
Default Value
Range
Example
Appendix tag in the Supported header. The caller then receives an UPDATE message from the callee, and displays the identity in the From header.
X ranges from 1 to 6.
Integer
0
0 to 2 account.1.cp_source = 0
Parameter- account.X.dtmf.type
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the DTMF type for account X.
If it is set to 0 (INBAND), DTMF digits are transmitted in the voice band.
If it is set to 1 (RFC 2833), DTMF digits are transmitted by RTP Events compliant to RFC
2833.
If it is set to 2 (SIP INFO), DTMF digits are transmitted by the SIP INFO messages.
If it is set to 3 (AUTO or SIP INFO), the IP phone negotiates with the other end to use
INBAND or RFC 2833, if there is no negotiation, using SIP INFO by default.
X ranges from 1 to 6.
Integer
1
Valid values are:
0-INBAND
1-RFC 2833
2-SIP INFO
3-AUTO or SIP INFO account.1.dtmf.type = 1
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Parameter- account.X.dtmf.dtmf_payload
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the RFC 2833 payload type.
X ranges from 1 to 6.
Integer
101
96 to 127 account.1.dtmf.dtmf_payload = 101
Parameter- account.X.dtmf.info_type
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the DTMF info type when the
DTMF type is configured as “SIP INFO”,
“AUTO or SIP INFO”.
X ranges from 1 to 6.
Integer
0
Valid values are:
0-Disabled
1-DTMF-Relay
2-DTMF
3-Telephone-Event account.1.dtmf.info_type = 0
Parameter- features.dtmf.repetition
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the number of times for the IP phone to send the end RTP EVENT packet.
Integer
3
1 to 3 features.dtmf.repetition = 3
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Appendix
Parameter- features.dtmf.hide
Description
Format
Default Value
Range
Example
Parameter- features.dtmf.hide_delay
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to suppress the display of DTMF digits.
If it is set to 1 (Enabled), the DTMF digits are displayed as asterisks.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.dtmf.hide = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to display the DTMF digits for a short period before displaying asterisks.
Note: It works only if the parameter
“features.dtmf.hide” is set to 1 (Enabled). It is not applicable to SIP-T20P IP phones.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.dtmf.hide_delay = 1
Parameter- features.dtmf.replace_tran
Description
Configuration File
<y0000000000xx>.cfg
Enables or disables transfer via DTMF
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Format
Default Value
Range
Example feature.
If it is set to 0 (Disabled), the IP phone will perform the transfer as normal when pressing the transfer key during a call.
If it is set to 1 (Enabled), the IP phone will transmit the specified DTMF digits to the server for completing call transfer when pressing the transfer key during a call.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.dtmf.replace_tran = 1
Parameter- features.dtmf.transfer
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the DTMF digits to be transmitted to complete the transfer.
Note: It works only if the parameter
“features.dtmf.replace_tran” is set to 1
(Enabled).
String
Blank
Valid values are: 0-9, *, # and A-D.
String within 32 characters features.dtmf.transfer = 123
332
Parameter- features.intercom.allow
Description
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to automatically answer an incoming intercom call.
If it is set to 0 (Disabled), the IP phone will reject incoming intercom calls and sends a
Format
Default Value
Range
Example
Parameter- features.intercom.mute
Description
Format
Default Value
Range
Example
Parameter- features.intercom.tone
Description
Appendix busy signal to the caller.
If it is set to 1 (Enabled), the IP phone will automatically answer an incoming intercom call.
Boolean
1
Valid values are:
0-Disabled
1-Enabled features.intercom.allow = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to mute the microphone when answering an intercom call.
If it is set to 0 (Disabled), the microphone is un-muted for incoming calls.
If it is set to 1 (Enabled), the microphone is muted for intercom calls.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.intercom.mute = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to play a warning tone when receiving an intercom call.
If it is set to 0 (Disabled), the IP phone will automatically answer the intercom call without a warning tone.
If it is set to 1 (Enabled), the IP phone will
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Format
Default Value
Range
Example play a warning tone to alert you before answering the intercom call.
Boolean
1
Valid values are:
0-Disabled
1-Enabled features.intercom.tone = 1
Parameter- features.intercom.barge
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to automatically answer an incoming intercom call while there is already an active call on the IP phone.
If it is set to 0 (Disabled), the IP phone will handle an incoming intercom call like a waiting call while there is already an active call on the IP phone.
If it is set to 1 (Enabled), the IP phone will automatically answer the intercom call while there is already an active call on the IP phone and place the active call on hold.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.intercom.barge = 1
334
Parameter- features.alert_info_tone
Description
Configuration File
<y0000000000xx>.cfg
Enables and disables the IP phone to map the keywords in the Alert-info header to the specified Bellcore ring tones.
Format
Default Value
Range
Appendix
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.alert_info_tone = 1 Example
Parameter- account.X.alert_info_url_enable
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Enables or disables distinctive ring tones feature for account X.
If it is set to 1 (Enabled), the IP phone will try to download the WAV ring tone file from the
URL and then play the remote ring tone when the Alert-Info header contains a remote URL.
X ranges from 1 to 6.
Boolean
1
Valid values are:
0-Disabled
1-Enabled account.1.alert_info_url_enable = 1
Parameter- distinctive_ring_tones.alert_info.X
.text
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures the texts to map the keywords contained in the SIP header.
X ranges from 1 to 10.
String
Blank
String within 32 characters distinctive_ring_tones.alert_info.1.text = family
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Parameter- distinctive_ring_tones.alert_info.X
.ringer
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures the desired ring tones for each text.
The value ranges from 1 to 5, the digit stands for the appropriate ring tone.
X ranges from 1 to 10.
Integer
1
Valid values are:
1-Ring1.wav
2-Ring2.wav
3-Ring3.wav
4-Ring4.wav
5-Ring5.wav distinctive_ring_tones.alert_info.1.ringer =
1
Parameter- voice.tone.country
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Configures the country tone for the IP phone.
String
Custom
Valid values are:
Custom
Australia
Austria
Brazil
Belgium
China
Czech
Denmark
Finland
France
336
Example
Appendix
Germany
Great Britain
Greece
Hungary
Lithuania
India
Italy
Japan
Mexico
New Zealand
Netherlands
Norway
Portugal
Spain
Switzerland
Sweden
Russia
United States
Chile
Czech ETSI
voice.tone.country = Custom
Parameter- voice.tone.dial voice.tone.ring voice.tone.busy voice.tone.congestion voice.tone.callwaiting voice.tone.dialrecall voice.tone.info voice.tone.stutter voice.tone.message (not applicable to SIP-T20P IP phones) voice.tone.autoanswer
Configuration File
<y0000000000xx>.cfg
Description
Configures the tone for each condition. tonelist = element[,element] [,element]…
Where element = [!]Freq1[+Freq2][+Freq3][+Freq4]
/Duration
Freq: the frequency of the tone (ranges from
200 to 7000 Hz). If it is set to 0 (0Hz), it means the tone is not played. A tone is comprised of
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Format
Default Value
Range
Example at most four different frequencies.
Duration: the time duration (in milliseconds, ranges from 0 to 30000ms) of the ring tone.
You can configure at most eight different tones for one condition, and separate tones by commas (e.g., 250/200, !0/1000,
200+300/500, 600+700+800+1000/2000). The exclamation point (!) can be added optionally, which means these tones are only played once.
Note: It works only if the parameter
“voice.tone.country” is set to Custom.
Refer to the introduction above
Blank
Not Applicable voice.tone.dial = 800+200/1000, 0/100,
500/1200, 500+600+950+1500/5000
Parameter- remote_phonebook.data.X.url
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the remote
XML phone book.
X ranges from 1 to 5.
Note: It is not applicable to SIP-T20P IP phones.
URL
Blank
String within 511 characters remote_phonebook.data.1.url = http://192.168.1.20/phonebook.xml
Parameter- remote_phonebook.data.X.name
Description
Configuration File
<y0000000000xx>.cfg
Configures the display name of the remote phone book item.
338
Format
Default Value
Range
Example
String
Blank
Appendix
Note: It is not applicable to SIP-T20P IP phones.
String within 99 characters remote_phonebook.data.1.name = yl01
Parameter- remote_phonebook.display_name
Description
Configuration File
<y0000000000xx>.cfg
Configures the display name of the remote phone book.
If you leave it blank, Remote Phone Book is displayed on the LCD screen at the path
Menu->Directory.
Note: It is not applicable to SIP-T20P IP phones.
String Format
Default Value
Range
Example
Blank
String within 99 characters remote_phonebook.display_name =
Remote Phone Book
Parameter- features.remote_phonebook.enabl
e
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Enables or disables the IP phone to perform a remote phone book search when receiving an incoming call.
Note: It is not applicable to SIP-T20P IP phones.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.remote_phonebook.enable = 1
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Parameter- features.remote_phonebook.flash_
time
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures how often to refresh the local cache of the remote phone book.
If it is set to 3600 (3600s), the IP phone will refresh the local cache of the remote phone book every 3600 seconds.
Note: It is not applicable to SIP-T20P IP phones.
Integer
21600
3600 to 2592000 features.remote_phonebook.flash_time =
21600
Configuration File
<y0000000000xx>.cfg
Enables or disables LDAP feature on the IP phone.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
Boolean
0
Valid values are:
0-Disabled
1-Enabled ldap.enable =1
Configuration File
<y0000000000xx>.cfg
Configures the name attribute for LDAP searching. The “*” symbol in the filter stands for any character. The “%” symbol in the filter
340
Parameter- ldap.enable
Description
Format
Default Value
Range
Example
Parameter- ldap.name_filter
Description
Format
Default Value
Range
Example
Parameter- ldap.number_filter
Description
Format
Default Value
Range
Example
Appendix stands for the entering string used as the prefix of the filter condition.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
String
Blank
String within 99 characters ldap.name_filter = (|(cn=%)(sn=%))
When the name prefix of the cn or sn of the contact record matches the search criteria, the record will be displayed on the LCD screen.
Configuration File
<y0000000000xx>.cfg
Configures the number attribute for LDAP searching.
The “*” symbol in the filter stands for any character. The “%” symbol in the filter stands for the entering string used as the prefix of the filter condition.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
String
Blank
String within 99 characters ldap.number_filter =
(|(telephoneNumber=%)(Mobile=%)(ipPh one=%))
When the number prefix of the telephoneNumber, Mobile or ipPhone of the contact record matches the search criteria, the record will be displayed on the
LCD screen.
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Parameter- ldap.host
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the IP address or domain name of the LDAP server.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
IP Address or Domain Name
Blank
String within 99 characters ldap.host = 192.168.1.20
Parameter- ldap.port
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the LDAP server port.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
Integer
389
1 to 65535 ldap.port = 389
Parameter- ldap.base
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the LDAP search base which corresponds to the location in the LDAP phone book from which the LDAP search request begins. The search base narrows the search scope and decreases directory search time.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
String
Blank
String within 99 characters ldap.base = dc=yealink,dc=cn
342
Parameter- ldap.user
Description
Format
Default Value
Range
Example
Parameter- ldap.password
Description
Format
Default Value
Range
Example
Parameter- ldap.max_hits
Description
Appendix
Configuration File
<y0000000000xx>.cfg
Configures the user name uses to login the
LDAP server.
This parameter can be left blank in case the server allows anonymous to login. Otherwise you will need to provide the user name to access the LDAP server.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
String
Blank
String within 99 characters ldap.user = cn=manager,dc=yealink,dc=cn
Configuration File
<y0000000000xx>.cfg
Configures the password to login the LDAP server.
This parameter can be left blank in case the server allows anonymous to login. Otherwise you will need to provide the password to access the LDAP server.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
String
Blank
String within 99 characters ldap.password = secret
Configuration File
<y0000000000xx>.cfg
Configures the maximum number of search results to be returned by the LDAP server. If the value of the “Max.Hits” is blank, the
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Format
Default Value
Range
Example
LDAP server will return all searched results.
Please note that a very large value of the
“Max. Hits” will slow down the LDAP search speed, therefore it should be configured according to the available bandwidth.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
Integer
50
1 to 32000 ldap.max_hits = 50
Parameter- ldap.name_attr
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the name attributes of each record to be returned by the LDAP server. It compresses the search results. You can configure multiple name attributes separated by spaces.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
String
Blank
String within 99 characters ldap.name_attr = cn sn
Parameter- ldap.numb_attr
Description
Format
Default Value
Configuration File
<y0000000000xx>.cfg
Configures the number attributes of each record to be returned by the LDAP server. It compresses the search results. You can configure multiple number attributes separated by spaces.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
String
Blank
344
Range
Example
Parameter- ldap.display_name
Description
Format
Default Value
Range
Example
Parameter- ldap.version
Description
Format
Default Value
Range
Example
Parameter- ldap.call_in_lookup
Description
Appendix
String within 99 characters ldap.numb_attr = telephoneNumber
Configuration File
<y0000000000xx>.cfg
Configures the display name of the contact record displayed on the LCD screen.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones. The value must start with
“%” symbol.
String
Blank
String within 99 characters ldap.display_name = %cn
The cn of the contact record is displayed on the LCD screen.
Configuration File
<y0000000000xx>.cfg
Configures the LDAP protocol version supported by the IP phone. Make sure the protocol value corresponds with the version assigned on the LDAP server.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
Integer
3
2 or 3 ldap.version = 3
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to perform an LDAP search when receiving an incoming call.
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Format
Default Value
Range
Example
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
Boolean
0
Valid values are:
0-Disabled
1-Enabled ldap.call_in_lookup = 1
Parameter- ldap.ldap_sort
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to sort the search results in alphabetical order or numerical order.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
Boolean
0
Valid values are:
0-Disabled
1-Enabled ldap.ldap_sort = 1
346
Visual and Audio Alert for BLF Pickup
Parameter- features.pickup.blf_visual_enabl
e
Configuration File
<MAC>.cfg
Description
Format
Enables or disables the IP phone to display a visual prompt when the monitored user receives an incoming call.
Note: It is not applicable to SIP-T19P and
SIP-T20P IP phones.
Boolean
Default Value
Range
Appendix
0
Valid values are:
0-Disabled
1-Enabled features.pickup.blf_visual_enable = 1 Example
Parameter- features.pickup.blf_audio_enable
Configuration File
<MAC>.cfg
Description
Format
Default Value
Enables or disables the IP phone to play an alert tone when the monitored user receives an incoming call.
Note: It is not applicable to SIP-T19P IP phones.
Boolean
Range
0
Valid values are:
0-Disabled
1-Enabled
Example features.pickup.blf_audio_enable = 1
BLF LED Mode
Parameter- features.blf_led_mode
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
It configures BLF LED mode and provides four kinds of definition for the BLF key LED status.
For more information, refer to Busy Lamp
Note: It is not applicable to SIP-T19P IP phones.
Integer
0
0 to 3 features.blf_led_mode = 1
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Parameter- account.X.music_server_uri
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the Music on Hold server address. Examples for valid values:
<10.1.3.165>, 10.1.3.165, sip:[email protected],
<sip:[email protected]>, <yealink.com> or yealink.com.
X ranges from 1 to 6.
Note: The DNS query in this parameter only supports A query.
String
Blank
String within 256 characters account.1.music_server_uri =<10.1.3.165>
Parameter- account.X.acd.enable
Description
Format
Default Value
Value
Example
Parameter- account.X.acd.available
Description
348
Configuration File
<MAC>.cfg
Enables or disables ACD feature for account
X.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.acd.enable = 1
Configuration File
<MAC>.cfg
Enables or disables the IP phone to display the available and unavailable soft keys after the phone logs into the ACD system for
Format
Default Value
Value
Example
Parameter- acd.auto_available
Description
Format
Default Value
Value
Example
Parameter- acd.auto_available_timer
Description
Format
Default Value
Value
Example
Appendix account X.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.acd.available = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables ACD auto available feature.
If it is set to 1 (Enabled), the IP phone will automatically change the phone status to available.
Boolean
0
Valid values are:
0-Disabled
1-Enabled acd.auto_available = 1
Configuration File
<y0000000000xx>.cfg
Configures the length of time (in seconds) before the IP phone state is automatically changed to available.
Note: It works only if the parameter
“acd.auto_available” is set to 1 (Enabled).
Integer
60
0 to 120 acd.auto_available_timer = 60
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Parameter- account.X.subscribe_mwi
Description
Format
Default Value
Value
Example
Configuration File
<MAC>.cfg
Enables or disables the IP phone to subscribe the message waiting indicator to the account for account X.
If it is set to 1 (Enabled), the IP phone will send a SUBSCRIBE message to the server for message-summary updates.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.subscribe_mwi = 0
Parameter- account.X.subscribe_mwi_expires
Description
Format
Default Value
Value
Example
Configuration File
<MAC>.cfg
Configures MWI subscribe expiry time (in seconds) for account X.
The IP phone is able to successfully refresh the SUBCRIBE for message-summary events before expiration of the SUBSCRIBE dialog.
X ranges from 1 to 6.
Note: It works only if the parameter
“account.X.subscribe_mwi” is set to 1
(Enabled).
Integer
3600
0 to 84600 account.1.subscribe_mwi_expires = 3600
350
Parameter- voice_mail.number.X
Description
Format
Default Value
Value
Example
String
Blank
Appendix
Configuration File
<MAC>.cfg
Configures the voice mail number for account X.
X ranges from 1 to 6.
String within 99 characters voice_mail.number.1 = 1234
Parameter- account.X.subscribe_mwi_to_vm
Description
Format
Default Value
Value
Example
Configuration File
<MAC>.cfg
Enables or disables the IP phone to subscribe the message waiting indicator to the voice mail number for account X.
X ranges from 1 to 6.
Note: It works only if the parameters
“account.X.subscribe_mwi” is set to 1
(Enabled) and “voice_mail.number.X” is configured.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.subscribe_mwi_to_vm = 0
Parameter- multicast.codec
Description
Format
Configuration File
<y0000000000xx>.cfg
Configures a multicast codec for the IP phone to use to send an RTP stream.
Note: It is not applicable to SIP-T19P IP phones. string
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Default Value
Range
Example
G722
Valid values are:
PCMU
PCMA
G729
G722
G726-16 (not applicable to SIP-T21P)
G726-24 (not applicable to SIP-T21P)
G726-32
G726-40 (not applicable to SIP-T21P)
G723_53 multicast.codec = G722
Parameter- multicast.receive_priority.enable
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to handle the incoming multicast paging calls when there is an active multicast paging call on the IP phone.
If it is set to 1 (Enabled), the IP phone will answer the incoming multicast paging call with a higher priority and ignore that with a lower priority.
Boolean
1
Valid values are:
0-Disabled
1-Enabled multicast.receive_priority.enable =1
Parameter- multicast.receive_priority.priority
Description
Configuration File
< y0000000000xx >.cfg
Configures the priority of multicast paging calls.
352
Format
Default Value
Range
Example
Appendix
1 is the highest priority, 10 is the lowest priority.
If it is set to 0, all incoming multicast paging calls will be automatically ignored.
Integer
10
0 to10 multicast.receive_priority.priority = 10
Parameter- multicast.listen_address.X.label
Description
Format
Default Value
Range
Example
Configuration File
< y0000000000xx >.cfg
Configures the label to be displayed on the
LCD screen when receiving the RTP multicast.
X ranges from 1 to 10.
String
Blank
String within 99 characters multicast.listen_address.1.label = Paging1
Parameter- multicast.listen_address.X.ip_addr
ess
Configuration File
< y0000000000xx >.cfg
Description
Format
Default Value
Range
Example
Configures the multicast address and port number that the IP phone listens to.
X ranges from 1 to 10.
Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
String
Blank
Not Applicable multicast.listen_address.1.ip_address =
224.5.6.20:10008
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354
Parameter- action_url.setup_completed action_url.registered action_url.unregistered action_url.register_failed action_url.off_hook action_url.on_hook action_url.incoming_call action_url.outgoing_call action_url.call_established action_url.dnd_on action_url.dnd_off action_url.always_fwd_on action_url.always_fwd_off action_url.busy_fwd_on action_url.busy_fwd_off action_url.no_answer_fwd_on action_url.no_answer_fwd_off action_url.transfer_call action_url.blind_transfer_call action_url.attended_transfer_call action_url.hold action_url.unhold action_url.mute action_url.unmute action_url.missed_call action_url.call_terminated action_url.busy_to_idle action_url.idle_to_busy action_url.ip_change action_url.forward_incoming_call action_url.reject_incoming_call action_url.answer_new_incoming_
call action_url.transfer_finished
Configuration File
<y0000000000xx>.cfg
action_url.transfer_failed
Description
Format
Default Value
Range
Example
Appendix
Configures the URL for the predefined event.
The value format is: http(s)://IP address of server/help.xml? variable name=variable value.
Valid variable values are:
$mac
$ip
$model
$firmware
$active_url
$active_user
$active_host
$local
$remote
$display_local
$display_remote
$call_id
URL
Blank
String within 511 characters action_url.mute = http://192.168.0.20/help.xml?model=$mo del
Parameter- features.action_uri_limit_ip
Description
Configuration File
<y0000000000xx>.cfg
Configures the address(es) from which
Action URI will be accepted.
For discontinuous IP addresses, multiple IP addresses are separated by commas.
For continuous IP addresses, the format likes
*.*.*.* and the “*” stands for the values
0~255.
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Format
Default Value
Range
Example
For example: 10.10.*.* stands for the IP addresses that range from 10.10.0.0 to
10.10.255.255.
If left blank, the IP phone cannot receive or handle any HTTP GET request.
If it is set to “any”, the IP phone will accept and handle HTTP GET requests from any IP address.
IP Address or any
Blank
String within 511 characters features.action_uri_limit_ip = any
Parameter- account.X.sip_server.Y.address
Description
Format
Default Value
Range
Example
Parameter- account.X.sip_server.Y.port
Description
Format
Default Value
Configuration File
<MAC>.cfg
Configures the IP address or domain name of the SIP server Y for account X.
X ranges from 1 to 6.
Y ranges from 1 to 2.
IP Address or Domain Name
Blank
String within 256 characters account.1.sip_server.1.address = yealink.pbx.com
Configuration File
<MAC>.cfg
Configures the port of the SIP server Y for account X.
X ranges from 1 to 6.
Y ranges from 1 to 2.
Integer
5060
356
Range
Example
Parameter- account.X.sip_server.Y.expires
Description
Appendix
0 to 65535 account.1.sip_server.1.port = 5060
Configuration File
<MAC>.cfg
Configures the registration expires (in seconds) of the SIP server Y for account X.
X ranges from 1 to 6.
Y ranges from 1 to 2.
Integer
3600
30 to 2147483647 account.1.sip_server.1.expires = 3600
Format
Default Value
Range
Example
Parameter- account.X.sip_server.Y.retry_counts
Configuration File
<MAC>.cfg
Description
Configures the retry times for the IP phone to resend requests when the SIP server Y does not respond correctly for account X.
X ranges from 1 to 6.
Y ranges from 1 to 2.
Format
Default Value
Range
Example
Integer
3
0 to 20 account.1.sip_server.1.retry_counts = 3
Fallback Mode
Parameter- account.X.fallback.redundancy_ty
pe
Configuration File
<MAC>.cfg
Description
Format
Default Value
Configures the registration mode for the IP phone in fallback mode.
X ranges from 1 to 6.
Integer
0
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Range
Valid values are:
0-Concurrent registration
1-Successive registration account.1.fallback.redundancy_type = 0 Example
Parameter- account.X.fallback.timeout
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the time interval (in seconds) for the IP phone to detect whether the working server is available by sending the registration request after the fallback server takes over call control.
It is only applicable to successive registration mode.
X ranges from 1 to 6.
Integer
120
10 to 2147483647 account.1.fallback.timeout = 120
Failover Mode
Parameter- account.X.sip_server.Y.failback_mo
de
Configuration File
<MAC>.cfg
Description
Format
Default Value
Range
Configures the mode for the IP phone to retry the primary server in failover mode.
X ranges from 1 to 6.
Y ranges from 1 to 2.
Integer
0
Valid values are:
0-newRequests: all requests are sent to the primary server first, regardless of the last used server.
1-DNSTTL: the IP phone retries to send requests to the primary server after the
358
Example
Appendix timeout equal to the DNSTTL configured for the server that the IP phone is registered to.
2-registration: the IP phone retries to send
REGISTER requests to the primary server when registration renewal.
3-duration: the IP phone retries to send requests to the primary server after the timeout defined by the account.X.sip_server.Y.failback_timeout parameter. account.1.sip_server.1.failback_mode =
0
Parameter- account.X.sip_server.Y.failback_tim
eout
Configuration File
<MAC>.cfg
Description
Format
Default Value
Range
Example
Configures the timeout (in seconds) for the IP phone to retry to send requests to the primary server after failing over to the current working server when the parameter
“account.X.sip_server.Y.failback_mode” is set to 3 (duration).
If you set the parameter to 0, the IP phone will not send requests to the primary server until a failover event occurs with the current working server.
X ranges from 1 to 6.
Y ranges from 1 to 2.
Integer
3600
0, 60 to 65535 account.1.sip_server.1.failback_timeout =
3600
Parameter- account.X.sip_server.Y.register_on_
enable
Configuration File
<MAC>.cfg
Description
Enables or disables the IP phone to register to the secondary server before sending
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Format
Default Value
Range
Example requests to the secondary server in the failover mode.
X ranges from 1 to 6.
Y ranges from 1 to 2.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.sip_server.1.register_on_enable
= 0
Parameter- account.X.transport
Description
Format
Default Value
Range
Example
Parameter- account.X.naptr_build
Description
360
Configuration File
<MAC>.cfg
Configures the transport type for account X.
If the parameter is set to 3 (DNS-NAPTR) and no server port is given, the IP phone performs the DNS NAPTR and SRV queries for the service type and port.
X ranges from 1 to 6.
Integer
0
Valid values are:
0-UDP
1-TCP
2-TLS
3-DNS-NAPTR account.1.transport = 3
Configuration File
<MAC>.cfg
Configures UDP SRV query or TCP/TLS SRV query for the IP phone to be performed when no result is returned from NAPTR
Format
Default Value
Range
Example
Appendix query.
X ranges from 1 to 6.
Integer
0
Valid values are:
0-UDP
1-TCP or TLS. account.1.naptr_build = 0
Parameter- network.lldp.enable
Description
Format
Default Value
Range
Example
Parameter- network.lldp.packet_interval
Description
Format
Default Value
Configuration File
<y0000000000xx>.cfg
Enables or disables LLDP feature on the IP phone.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
1
Valid values are:
0-Disabled
1-Enabled network.lldp.enable = 1
Configuration File
<y0000000000xx>.cfg
Configures the amount of time (in seconds) between the transmissions of LLDP packet.
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It works only if the parameter
“network.lldp.enable” is set to 1 (Enabled).
Integer
60
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Range
Example
1 to 3600 network.lldp.packet_interval = 60
Internet Port
Parameter- network.vlan.internet_port_enable
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Enables or disables the IP phone to insert
VLAN tag on packet from the Internet port.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
Range
0
Valid values are:
0-Disabled
1-Enabled
Example network.vlan.internet_port_enable = 1
Parameter- network.vlan.internet_port_vid
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the VLAN ID that is associated with the particular VLAN.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
1
1 to 4094 network.vlan.internet_port_vid = 1
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Appendix
Parameter- network.vlan.internet_port_priority
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the priority value used for passing VLAN packets.
7 is the highest priority, 0 is the lowest priority.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
0 to 7 network.vlan.internet_port_priority = 0
PC Port
Parameter- network.vlan.pc_port_enable
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to insert
VLAN tag on packet from the PC port.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
0
Valid values are:
0-Disabled
1-Enabled network.vlan.pc_port_enable = 1
Parameter- network.vlan.pc_port_vid
Description
Configuration File
<y0000000000xx>.cfg
Configures the VLAN ID that is associated with the particular VLAN.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Format
Default Value
Range
Example
Integer
1
1 to 4094 network.vlan.pc_port_vid = 1
Parameter- network.vlan.pc_port_priority
Description
Format
Default Value
Range
Example
DHCP VLAN Discovery
Parameter- network.vlan.dhcp_enable
Configuration File
<y0000000000xx>.cfg
Configures the priority value used for passing VLAN packets.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
0 to 7 network.vlan.pc_port_priority = 0
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables DHCP VLAN discovery feature on the IP phone.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
1
Valid values are:
0-Disabled
1-Enabled network.vlan.dhcp_enable = 1
Parameter- network.vlan.dhcp_option
Description
Configuration File
<y0000000000xx>.cfg
Configures the DHCP option used to
364
Format
Default Value
Range
Example
Parameter- network.vpn_enable
Description
Format
Default Value
Range
Example
Parameter- openvpn.url
Description
Format
Default Value
Range
Appendix request the VLAN ID.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
132
128 to 254 network.vlan.dhcp_option = 132
Configuration File
<y0000000000xx>.cfg
Enables or disables VPN feature on the IP phone.
Note: It is not applicable to SIP-T19P IP phones. If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
0
Valid values are:
0-Disabled
1-Enabled network.vpn_enable = 1
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the OpenVPN
TAR package.
Note: It is not applicable to SIP-T19P IP phones.
URL
Blank
String within 511 characters
365
Parameter- network.qos.rtptos
Description
Format
Default Value
Range
Example
Parameter- network.qos.signaltos
Description
Format
Default Value
Range
Example
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Example openvpn.url = http://192.168.10.25/OpenVPN.tar
Configuration File
<y0000000000xx>.cfg
Configures the DSCP for voice packets.
The default DSCP value for RTP packets is
46 (Expedited Forwarding).
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
46
0 to 63 network.qos.rtptos = 46
Configuration File
<y0000000000xx>.cfg
Configures the DSCP for SIP packets.
The default DSCP value for SIP packets is 26
(Assured Forwarding).
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
26
0 to 63 network.qos.signaltos = 26
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Appendix
Parameter- account.X.nat.nat_traversal
Description
Format
Default Value
Range
Example
Parameter- account.X.nat.stun_server
Description
Format
Default Value
Range
Example
Parameter- account.X.nat.stun_port
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Enables or disables the NAT traversal for account X.
X ranges from 1 to 6.
Boolean
0
Valid values are:
0-Disabled
1-Enabled account.1.nat.nat_traversal = 0
Configuration File
<MAC>.cfg
Configures the IP address or the domain name of the STUN server for account X.
X ranges from 1 to 6.
IP Address or Domain Name
Blank
String within 99 characters account.1.nat.stun_server =
218.107.220.201
Configuration File
<MAC>.cfg
Configures the port of the STUN server.
X ranges from 1 to 6.
Integer
3478
1024 to 65000 account.1.nat.stun_port = 3478
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368
Parameter- network.802_1x.mode
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Configures the types of the 802.1X authentication to use on the IP phone.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
Valid values are:
0-Disabled
1-EAP-MD5
2-EAP-TLS
3-PEAP-MSCHAPv2
4-EAP-TTLS/EAP-MSCHAPv2 network.802_1x.mode = 1 Example
Parameter- network.802_1x.identity
Description
Format
Default Value
Range
Example
Parameter- network.802_1x.md5_password
Description
Configuration File
<y0000000000xx>.cfg
Configures the identity used for authenticating the IP phone.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
String
Blank
String within 32 characters network.802_1x.identity = admin
Configuration File
<y0000000000xx>.cfg
Configures the password used for authenticating the IP phone.
Format
Default Value
Range
Example
Parameter- network.802_1x.root_cert_url
Description
Format
Default Value
Range
Example
Appendix
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is only applicable to EAP-MD5,
PEAP-MSCHAPv2 and
EAP-TTLS/EAP-MSCHAPv2 protocols.
String
Blank
String within 32 characters network.802_1x.md5_password = admin123
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the CA certificate used for authentication.
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is only applicable to EAP-TLS,
PEAP-MSCHAPv2 and
EAP-TTLS/EAP-MSCHAPv2 protocols. The format of the certificate must be *.pem,
*.crt, *.cer or *.der.
URL
Blank
String within 511 characters network.802_1x.root_cert_url = http://192.168.1.10/ca.pem
Parameter- network.802_1x.client_cert_url
Description
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the device certificate used for authentication.
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is only applicable to the EAP-TLS protocol. The format of the certificate must be *.pem or *.cer.
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Format
Default Value
Range
Example
URL
Blank
String within 511 characters network.802_1x.client_cert_url = http://192.168.1.10/ client.pem
Parameter- managementserver.enable
Description
Format
Default Value
Range
Example
Parameter- managementserver.username
Description
Configuration File
<y0000000000xx>.cfg
Enables or disables TR-069 feature on the IP phone.
Integer
0
Valid values are:
0-Disabled
1-Enabled managementserver.enable = 1
Format
Default Value
Range
Example
Parameter- managementserver.password
Description
Configuration File
<y0000000000xx>.cfg
Configures the user name to authenticate with the ACS. This string is set to the empty string if no authentication is required.
String
Blank
String within 128 characters managementserver.username = user1
Configuration File
<y0000000000xx>.cfg
Configures the password to authenticate with the ACS. This string is set to the empty string if no authentication is required.
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Appendix
Format
Default Value
Range
Example
Parameter- managementserver.url
Description
Format
Default Value
Range
Example
String
Blank
String within 64 characters managementserver.password = pwd123
Configuration File
<y0000000000xx>.cfg
Configures the URL of the ACS.
URL
Blank
String within 511 characters managementserver.url = http://192.168.1.20/acs/
Parameter- managementserver.connection_re
quest_username
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures the user name for the IP phone to authenticate the incoming connection requests.
String
Blank
String within 128 characters managementserver.connection_request_
username = acsuser
Parameter- managementserver.connection_re
quest_password
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Configures the password for the IP phone to authenticate the incoming connection requests.
String
Blank
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Range
Example
String within 64 characters managementserver.connection_request_
password = acspwd
Parameter- managementserver.periodic_infor
m_enable
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Enables or disables the IP phone to periodically report its configuration information to the ACS.
Boolean
1
Valid values are:
0-Disabled
1-Enabled managementserver.periodic_inform_ena
ble = 1
Parameter- managementserver.periodic_infor
m_interval
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures the interval (in seconds) to report its configuration information to the
ACS.
Integer
60
5 to 4294967295 managementserver.periodic_inform_inte
rval = 60
372
Parameter- network.ip_address_mode
Description
Configuration File
<MAC>.cfg
Configures the IP address mode.
Note: If you change this parameter, the IP
Format
Default Value
Range
Example
Appendix phone will reboot to make the change take effect.
Integer
0
Valid values are:
0-IPv4
1-IPv6
2-IPv4&IPv6 network.ip_address_mode = 1
Parameter- network.ipv6_internet_port.type
Description
Format
Default Value
Range
Example
Parameter- network.ipv6_static_dns_enable
Description
Format
Default Value
Configuration File
<MAC>.cfg
Configures the IPv6 address assignment method.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
Valid values are:
0-DHCP
1-Static IP Address network.ipv6_internet_port.type = 0
Configuration File
<y0000000000xx>.cfg
Enables or disables the phone to use manually configured static IPv6 DNS when the parameter
“network.ipv6_internet_port.type” is set to
0 (DHCP).
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
0
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Range
Example
Valid values are:
0-Disabled
1-Enabled network.ipv6_static_dns_enable= 0
Parameter- network.ipv6_internet_port.ip
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the IPv6 address when the
IPv6 address assignment method is configured as Static IP Address and the IP address mode is configured as IPv6 or
IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
IPv6 Address
Blank
Not Applicable network.ipv6_internet_port.ip =
2026:1234:1:1:215:65ff:fe1f:caa
Parameter- network.ipv6_prefix
Description
Format
Default Value
Range
Example
Configuration File
<MAC>.cfg
Configures the prefix of the IPv6 address when the IPv6 address assignment method is configured as Static IP Address and the IP address mode is configured as
IPv6 or IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
64
0 to 128 network.ipv6_prefix = 64
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Parameter- network.ipv6_internet_port.gateway
Description
Configuration File
<MAC>.cfg
Configures the gateway when the IPv6 address assignment method is configured as Static IP Address and the IP address mode is configured as IPv6 or
IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Format
Default Value
Range
Example
IPv6 Address
Blank
Not Applicable network.ipv6_internet_port.gateway =
3036:1:1:c3c7:c11c:5447:23a6:255
Parameter- network.ipv6_primary_dns
Description
Format
Default Value
Range
Example
Parameter- network.ipv6_secondary_dns
Description
Configuration File
<MAC>.cfg
Configures the primary DNS server when the IPv6 address assignment method is configured as Static IP Address and the IP address mode is configured as IPv6 or
IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
IPv6 Address
Blank
Not Applicable network.ipv6_primary_dns =
3036:1:1:c3c7: c11c:5447:23a6:256
Configuration File
<MAC>.cfg
Configures the secondary DNS server when the IPv6 address assignment method is configured as Static IP Address
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Format
Default Value
Range
Example and the IP address mode is configured as
IPv6 or IPv4&IPv6.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
IPv6 Address
Blank
Not Applicable network.ipv6_secondary_dns =
2026:1234:1:1:c3c7:c11c:5447:23a6
Parameter- features.headset_prior
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables headset prior feature.
If it is set to 1 (enabled), a user needs to press the HEADSET key to activate the headset mode. The headset mode will not be deactivated until the user presses the
HEADSET key again.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.headset_prior = 1
376
Parameter- features.headset_training
Description
Configuration File
<y0000000000xx>.cfg
Enables or disables dual headset feature.
If it is set to 1 (Enabled), users can use two
Format
Default Value
Range
Example
Appendix headsets on one phone. When the IP phone joins in a cal, the users with the headset connected to the headset jack have a full-duplex conversation, while the users with the headset connected to the handset jack are only allowed to listen to.
Note: It is not applicable to SIP-T19P and
SIP-T21P IP phones.
Boolean
0
Valid values are:
0-Disabled
1-Enabled features.headset_training = 1
Parameter- account.X.codec.Y.enable
Description
Format
Default Value
Configuration File
<MAC>.cfg
Enables or disables the IP phone to use the specific codec for account X.
X ranges from 1 to 6.
Y ranges from 1 to 11.
Boolean
For SIP-T20P/T22P/T26P/T28P IP phones:
When Y=1, the default value is 1;
When Y=2, the default value is 1;
When Y=3, the default value is 0;
When Y=4, the default value is 0;
When Y=5, the default value is 1;
When Y=6, the default value is 1;
When Y=7, the default value is 0;
When Y=8, the default value is 0;
When Y=9, the default value is 0;
When Y=10, the default value is 0;
When Y=11, the default value is 0.
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Range
Example
For SIP-T19P/T21P IP phones:
When Y=1, the default value is 1;
When Y=2, the default value is 1;
When Y=3, the default value is 0;
When Y=4, the default value is 0;
When Y=5, the default value is 1;
When Y=6, the default value is 1;
When Y=7, the default value is 0;
When Y=8, the default value is 0.
Valid values are:
0-Disabled
1-Enabled account.1.codec.1.enable = 1
Parameter- account.X.codec.Y.payload_type
Description
Format
Default Value
Configuration File
<MAC>.cfg
Configures the codec for account X to use.
X ranges from 1 to 6.
Y ranges from 1 to 11.
String
For SIP-T20P/T22P/T26P/T28P IP phones:
When Y=1, the default value is PCMU;
When Y=2, the default value is PCMA;
When Y=3, the default value is G723_53;
When Y=4, the default value is G723_63;
When Y=5, the default value is G729;
When Y=6, the default value is G722;
When Y=7, the default value is iLBC;
When Y=8, the default value is G726-16;
When Y=9, the default value is G726-24;
When Y=10, the default value is G726-32;
When Y=11, the default value is G726-40.
For SIP-T19P/T21P IP phones:
When Y=1, the default value is PCMU;
When Y=2, the default value is PCMA;
When Y=3, the default value is G723_53;
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Range
Example
Parameter- account.X.codec.Y.priority
Description
Format
Default Value
Appendix
When Y=4, the default value is G723_63;
When Y=5, the default value is G729;
When Y=6, the default value is G722;
When Y=7, the default value is iLBC;
When Y=8, the default value is G726-32.
Valid values are:
PCMU
PCMA
G729
G722
G723_53
G723_63
G726-16
G726-24
G726-32
G726-40 iLBC account.1.codec.1.payload_type =
PCMU
Configuration File
<MAC>.cfg
Configures the priority for the codec.
X ranges from 1 to 6.
Y ranges from 1 to 11.
Integer
For SIP-T20P/T22P/T26P/T28P IP phones:
When Y=1, the default value is 1;
When Y=2, the default value is 2;
When Y=3, the default value is 0;
When Y=4, the default value is 0;
When Y=5, the default value is 3;
When Y=6, the default value is 4;
When Y=7, the default value is 0;
When Y=8, the default value is 0;
When Y=9, the default value is 0;
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Range
Example
Parameter- account.X.codec.Y.rtpmap
When Y=10, the default value is 0;
When Y=11, the default value is 0.
For SIP-T19P/T21P IP phones:
When Y=1, the default value is 1;
When Y=2, the default value is 2;
When Y=3, the default value is 0;
When Y=4, the default value is 0;
When Y=5, the default value is 3;
When Y=6, the default value is 4;
When Y=7, the default value is 0;
When Y=8, the default value is 0.
0 to 10 account.1.codec.1.priority = 1
Description
Format
Default Value
Configuration File
<MAC>.cfg
Configures the rtpmap.
X ranges from 1 to 6.
Y ranges from 1 to 11.
Integer
For SIP-T20P/T22P/T26P/T28P IP phones:
When Y=1, the default value is 0;
When Y=2, the default value is 8;
When Y=3, the default value is 4;
When Y=4, the default value is 4;
When Y=5, the default value is 18;
When Y=6, the default value is 9;
When Y=7, the default value is 106;
When Y=8, the default value is 103;
When Y=9, the default value is 104;
When Y=10, the default value is 102;
When Y=11, the default value is 105.
For SIP-T19P/T21P IP phones:
When Y=1, the default value is 0;
When Y=2, the default value is 8;
380
Range
Example
Ptime
Parameter- account.X.ptime
Description
Format
Default Value
Range
Example
Appendix
When Y=3, the default value is 4;
When Y=4, the default value is 4;
When Y=5, the default value is 18;
When Y=6, the default value is 9;
When Y=7, the default value is 106;
When Y=8, the default value is 102.
0 to 127 account.1.codec.1.rtpmap = 0
Configuration File
<MAC>.cfg
Configures the ptime (in milliseconds) for the codec.
X ranges from 1 to 6.
Integer
20
Valid values are:
0 (Disabled)
10, 20, 30, 40, 50, 60 account.1.ptime = 20
Parameter- voice.echo_cancellation
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables AEC feature on the IP phone.
Boolean
1
Valid values are:
0-Disabled
1-Enabled voice.echo_cancellation = 1
381
Parameter- voice.cng
Description
Format
Default Value
Range
Example
382
Parameter- voice.jib.adaptive
Description
Format
Default Value
Range
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Parameter- voice.vad
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables VAD feature on the IP phone.
Boolean
0
Valid values are:
0-Disabled
1-Enabled voice.vad = 1
Configuration File
<y0000000000xx>.cfg
Enables or disables CNG feature on the IP phone.
Boolean
1
Valid values are:
0-Disabled
1-Enabled voice.cng = 1
Configuration File
<y0000000000xx>.cfg
Configures the type of jitter buffer.
Integer
1
Valid values are:
Example
Parameter- voice.jib.min
Description
Format
Default Value
Range
Example
Parameter- voice.jib.max
Description
Format
Default Value
Range
Example
Parameter- voice.jib.normal
Description
Format
Default Value
Appendix
0-Fixed
1-Adaptive voice.jib.adaptive = 1
Configuration File
<y0000000000xx>.cfg
Configures the minimum delay time for jitter buffer.
Note: It works only if the parameter
“voice.jib.adaptive” is set to 1 (Adaptive).
Integer
60
0 to 400 voice.jib.min = 60
Configuration File
<y0000000000xx>.cfg
Configures the maximum delay time for jitter buffer.
Note: It works only if the parameter
“voice.jib.adaptive” is set to 1 (Adaptive).
Integer
240
0 to 400 voice.jib.max = 300
Configuration File
<y0000000000xx>.cfg
Configures the fixed delay time for jitter buffer.
Note: It works only if the parameter
“voice.jib.adaptive” is set to 0 (Fixed).
Integer
120
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Range
Example
0 to 400 voice.jib.mormal = 120
Parameter- account.X.transport
Description
Format
Default Value
Range
Example
Parameter- security.trust_certificates
Description
Format
Default Value
Range
384
Configuration File
<MAC>.cfg
Configures the transport type for account X.
If it is set to 2 (TLS), the SIP message of this account will be encrypted after the successful TLS negotiation.
X ranges from 1 to 6.
Integer
0 (UDP)
Valid values are:
0-UDP
1-TCP
2-TLS
3-DNS-NAPTR account.1.transport = 2
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to authenticate the connecting server based on the trusted certificates list.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
1
Valid values are:
0-Disabled
Example
Parameter- security.ca_cert
Description
Format
Default Value
Range
Example
Parameter- security.cn_validation
Description
Format
Default Value
Range
Example
Appendix
1-Enabled security.trust_certificates = 1
Configuration File
<y0000000000xx>.cfg
Configures the type of certificates the IP phone used to authenticate the connecting server.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
2
Valid values are:
0-Default certificates
1-Custom certificates
2-All certificates security.ca_cert = 2
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to mandatorily validate the CommonName or
SubjectAltName of the certificate sent by the connecting server.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Boolean
0
Valid values are:
0-Disabled
1-Enabled security.cn_validation = 0
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Parameter- security.dev_cert
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000 xx>.cfg
Configures the type of certificates the IP phone sends for authentication.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
Valid values are:
0-Default certificates
1-Custom certificates security.dev_cert = 0
386
Parameter- trusted_certificates.url
Description
Format
Default Value
Range
Example
Parameter- server_certificates.url
Description
Format
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the certificate used to authenticate the connecting server.
Note: The certificate you want to upload must be in *.pem, *.crt, *.cer or *.der format.
URL
Blank
String within 511 characters trusted_certificates.url = http://192.168.1.20/tc.crt
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the certificate the IP phone sends for authentication.
Note: The certificate you want to upload must be in *.pem or *.cer format.
URL
Default Value
Range
Example
Appendix
Blank
String within 511 characters server_certificates.url = http://192.168.1.20/ca.pem
Parameter- account.X.srtp_encryption
Description
Format
Default Value
Value
Example
Configuration File
<MAC>.cfg
Configures whether to use voice encryption service.
If it is set to 1 (Optional), the IP phone will negotiate with the other IP phone what type of encryption to utilize for the session.
If it is set to 2 (Compulsory), the IP phone is forced to use SRTP during a call.
X ranges from 1 to 6.
Integer
0
Valid values are:
0-Disabled
1-Optional
2-Compulsory account.1.srtp_encryption = 0
Parameter- auto_provision.aes_key_in_file
Description
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to decrypt configuration files using the encrypted AES keys.
If it is set to 1 (Enabled), the IP phone will download <y0000000000xx_Security>.enc and <MAC_Security>.enc files during auto provisioning, and then decrypts these files into the plaintext keys (e.g., key2, key3) respectively using the phone built-in key
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Format
Default Value
Value
Example
(e.g., key1). The IP phone then decrypts the encrypted configuration files using corresponding key (e.g., key2, key3).
If it is set to 0 (Disabled), the IP phone will decrypt the encrypted configuration files using plaintext AES keys configured on the
IP phone.
Boolean
0
Valid values are:
0-Disabled
1-Enabled auto_provision.aes_key_in_file = 0
Parameter- auto_provision.aes_key_16.com
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the plaintext AES key which is used to decrypt the <y0000000000xx>.cfg file.
Note: It works only if the parameter
“auto_provision.aes_key_in_file” is set to 0
(Disabled).
String
Blank
16 characters and the supported characters contain: 0 ~ 9, A ~ Z, a ~ z auto_provision.aes_key_16.com =
0123456789abcdef
Parameter- auto_provision.aes_key_16.mac
Description
Format
Configuration File
<y0000000000xx>.cfg
Configures the plaintext AES key which is used to decrypt the <MAC>.cfg file.
Note: It works only if the parameter
“auto_provision.aes_key_in_file” is set to 0
(Disabled).
String
388
Default Value
Range
Example
Appendix
Blank
16 characters and the supported characters contain: 0 ~ 9, A ~ Z, a ~ z auto_provision.aes_key_16.mac =
0123456789abmins
Parameter- auto_provision.update_file_mode
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to update encrypted configuration settings only during auto provisioning.
Boolean
0
Valid values are:
0-Disabled
1-Enabled auto_provision.update_file_mode = 0
Parameter- firmware.url
Description
Format
Default Value
Range
Example
Parameter- auto_provision.power_on
Description
Configuration File
<y0000000000xx>.cfg
Configures the access URL of firmware.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
URL
Blank
String within 511 characters firmware.url = http://192.168.1.20/2.72.0.1.rom
Configuration File
<y0000000000xx>.cfg
Enables or disables the IP phone to perform
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Format
Default Value
Range
Example an auto provisioning process when powered on.
Boolean
1
Valid values are:
0-Disabled
1-Enabled auto_provision.power_on = 1
Parameter- auto_provision.repeat.enable
Description
Format
Default Value
Range
Example
Configuration File
< y0000000000xx >.cfg
Enables or disables the IP phone to check new configuration repeatedly.
Boolean
0
Valid values are:
0-Disabled
1-Enabled auto_provision.repeat.enable =0
Parameter- auto_provision.repeat.minutes
Description
Format
Default Value
Range
Example
Configuration File
< y0000000000xx >.cfg
Configures the interval (in minutes) for the
IP phone to check new configuration.
Note: It works only if the parameter
“auto_provision.repeat.enable” is set to
1(Enabled).
Integer
1440
1 to 43200 auto_provision.repeat.minutes = 1000
390
Parameter- auto_provision.weekly.enable
Description
Format
Default Value
Range
Example
Appendix
Configuration File
< y0000000000xx >.cfg
Enables or disables the IP phone to check new configuration weekly.
Boolean
0
Valid values are:
0-Disabled
1-Enabled auto_provision.weekly.enable =0
Parameter- auto_provision.weekly.begin_time
Description
Format
Default Value
Range
Example
Configuration File
< y0000000000xx >.cfg
Configures the begin time of the day for the
IP phone to check new configuration weekly.
Note: It works only if the parameter
“auto_provision.weekly.enable” is set to
1(Enabled).
Time
00:00
00:00 to 23:59 auto_provision.weekly.begin_time = 01:30
Parameter- auto_provision.weekly.end_time
Description
Format
Default Value
Range
Configuration File
< y0000000000xx >.cfg
Configures the end time of the day for the
IP phone to check new configuration weekly.
Note: It works only if the parameter
“auto_provision.weekly.enable” is set to
1(Enabled).
Time
00:00
00:00 to 23:59
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Example auto_provision.weekly.end_time = 21:30
Parameter- auto_provision.weekly.dayofweek
Configuration File
< y0000000000xx >.cfg
Description
Format
Default Value
Configures the days of the week for the IP phone to check new configuration weekly.
Integer
Range
Example
0123456
Valid values are:
0-Sunday
1-Monday
2-Tuesday
3-Wednesday
4-Thursday
5-Friday
6-Saturday auto_provision.weekly.dayofweek =
0123456
Parameter- dialplan_replace_rule.url
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the replace rule template.
URL
Blank
String within 511 characters dialplan_replace_rule.url = http://192.168.10.25/dialplan.xml
392
Parameter- dialplan_dialnow.url
Description
Format
Default Value
Range
Example
Appendix
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the dial-now template.
URL
Blank
String within 511 characters dialplan_dialnow.url = http://192.168.10.25/dialnow.xml
Parameter- custom_softkey_call_failed.url
Description
Format
Default Value
Range
Example
Parameter- custom_softkey_call_in.url
Description
Format
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the CallFailed state.
URL
Blank
String within 511 characters
The following example uses HTTP to download the CallFailed state file from the “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_call_failed.url = http://10.2.8.16:8080/XMLfiles/CallFailed.x
ml
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the CallIn state.
URL
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Default Value
Range
Example
Blank
String within 511 characters
The following example uses HTTP to download the CallIn state file from the
“XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_call_in.url = http://10.2.8.16:8080/XMLfiles/CallIn.xml
Parameter- custom_softkey_connecting.url
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Connecting state.
URL
Blank
String within 511 characters
The following example uses HTTP to download the Connecting state file from the “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_connecting.url = http://10.2.8.16:8080/XMLfiles/Connecting.
xml
Parameter- custom_softkey_dialing.url
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Dialing state.
URL
Blank
String within 511 characters
The following example uses HTTP to download the Dialing state file from the
“XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port.
394
Appendix custom_softkey_dialing.url = http://10.2.8.16:8080/XMLfiles/Dialing.xml
Parameter- custom_softkey_ring_back.url
Description
Format
Default Value
Range
Example
Parameter- custom_softkey_talking.url
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the RingBack state.
URL
Blank
String within 511 characters
The following example uses HTTP to download the RingBack state file from the “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_ring_back.url = http://10.2.8.16:8080/XMLfiles/RingBack.x
ml
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Talking state.
URL
Blank
String within 511 characters
The following example uses HTTP to download the Talking state file from the
“XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_talking.url = http://10.2.8.16:8080/XMLfiles/Talking.xml
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Parameter- local_contact.data.url
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the local contact file.
URL
Blank
String within 511 characters local_contact.data.url = http://192.168.10.25/contact.xml
Parameter- remote_phonebook.data.X.url
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the remote
XML phone book.
X ranges from 1 to 5.
URL
Blank
String within 511 characters remote_phonebook.data.1.url = http://192.168.1.20/phonebook.xml
396
Parameter- directory_setting.url
Description
Format
Default Value
Range
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the directory template.
URL
Blank
String within 511 characters
Example
Parameter- super_search.url
Description
Format
Default Value
Range
Example
Appendix directory_setting.url = http://192.168.1.20/favorite_setting.xml
Configuration File
<y0000000000xx>.cfg
Configures the access URL of the super search template.
URL
Blank
String within 511 characters super_search.url = http://192.168.1.20/super_search.xml
Parameter- syslog.mode
Description
Format
Default Value
Range
Example
Parameter- syslog.server
Description
Configuration File
<y0000000000xx>.cfg
Configures the syslog mode.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
0
0-Local
1-Server syslog.mode = 1
Configuration File
<y0000000000xx>.cfg
Configures the IP address or domain name of the syslog server where to export the log
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Format
Default Value
Range
Example files.
Note: It works only if the parameter
“syslog.mode” is set to 1 (Server). If you change this parameter, the IP phone will reboot to make the change take effect.
IP Address or Domain Name
Blank
String within 99 characters syslog.server = 192.168.1.50
Parameter- syslog.log_level
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Configures the severity level of the logs to be reported to a log file.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Integer
3
0 to 6 syslog.log_level = 3
Parameter- watch_dog.enable
Description
Format
Default Value
Range
Example
Configuration File
<y0000000000xx>.cfg
Enables or disables Watch Dog feature.
Boolean
1
Valid values are:
0-Disabled
1-Enabled watch_dog.enable = 1
398
Appendix
This section provides the DSS key parameters you can configure on IP phones. DSS key consists of memory key, line key and programable key. The following table lists the number of DSS keys you can configure for each phone model:
Phone Model
SIP-T28P
SIP-T26P
SIP-T22P
SIP-T21P
SIP-T20P
SIP-T19P
Line Key
6
3
3
2
2
/
Memory Key
10
10
/
/
/
/
Programable Key
14
14
13
11
9
11
Note The programable key takes effect only if the IP phone is idle.
DSS key can be assigned with various key features. The parameters of the DSS key are detailed in the following:
Parameter- memorykey.X.type
Parameter- linekey.X.type
Parameter- programablekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Configures key feature for the DSS key.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
For memory keys:
Valid types are:
N/A
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BLF
URL
Group Listening
XML Group
Group Pickup
Multicast Paging
Record
XML Browser
URL Record
LDAP
Prefix
Zero Touch
ACD
Local Group
Custom Button
Keypad Lock
Directory
For line keys:
Conference
Forward
Transfer
Hold
DND
Call Return
SMS
Directed Pickup
Call Park
DTMF
Voice Mail
Speed Dial
Intercom
Line
Valid types are:
Conference
Forward
Transfer
Hold
400
Appendix
DND
Call Return
SMS (not applicable to SIP-T20P IP phones)
Directed Pickup
Call Park
DTMF
Voice Mail
Speed Dial
Intercom
Line
BLF
Group Listening
XML Group (not applicable to SIP-T20P IP phones)
Group Pickup
Multicast Paging
Record
XML Browser
Hot Desking
URL Record
LDAP (not applicable to SIP-T20P IP phones)
Prefix
Zero Touch
ACD
Local Group
Custom Button
Keypad Lock
Directory
For programable keys:
Valid types are:
N/A
Forward
DND
Call Return
SMS (not applicable to SIP-T20P IP phones)
Directed Pickup
Spead Dial
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Format
Default Value
XML Group (not applicable to SIP-T19P IP phones)
Group Pickup
XML Browser
History
Menu
Switch Account (not applicable to SIP-T19P
IP phones)
New SMS (not applicable to SIP-T20P IP phones)
Status
Hot Desking (only applicable to SIP-T19P IP phones)
LDAP (not applicable to SIP-T19P and
SIP-T20P IP phones)
Prefix (not applicable to SIP-T20P IP phones)
Zero Touch
Local Directory
Local Group
XML Directory (not applicable to SIP-T20P IP phones)
Keypad Lock
Directory
Integer
For the memory key, the default value is 0 (N/A).
For the line key, the default value is 15 (Line).
For the programable key, when x=1, the default value is 28. when x=2, the default value is 61. when x=3, the default value is 5. when x=4, the default value is 30. when x=5, the default value is 28. when x=6, the default value is 29. when x=7, the default value is 31. when x=8, the default value is 31. when x=9, the default value is 33. when x=10/11/12/13, the default value is 0.
402
Range
Appendix when x=14, the default value is 2.
Valid values are:
0-N/A
1-Conference
2-Forward
3-Transfer
4-Hold
5-DND
7-Call Return
8-SMS
9Directed Pickup
10-Call Park
11-DTMF
12-Voice Mail
13-Speed Dial
14-Intercom
15-Line
16-BLF
17-URL
18-Group Listening
22-XML Group
23-Group Pickup
24-Multicast Paging
25-Record
27-XML Browser
28-History
30-Menu
31-Switch Account
32-New SMS
33-Status
34-Hot Desking
35-URL Record
38-LDAP
40-Prefix
41-Zero Touch
42-ACD
43-Local Directory
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Example
45-Local Group
47-XML Directory
49-Custom Button
50-Keypad Lock
61-Directory memorykey.1.type = 8
Parameter- memorykey.X.line
Parameter- linekey.X.line
Parameter- programablekey.X.line
Configuration File
<y0000000000xx>.cfg
Description
Configures the desired line to apply the key feature.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
When assigning the following features, you do not need to configure this parameter:
DTMF
Prefix
XML Browser
LDAP (not applicable to SIP-T19P and
SIP-T20P)
Conference
Forward
Hold
DND
Call Return
SMS (not applicable to SIP-T20P)
Record
404
Format
Default Value
Range
Example
Parameter- memorykey.X.value
Parameter- linekey.X.value
Parameter- programablekey.X.value
Description
Appendix
URL Record
Multicast Paging
Group Listening
Local Group
XML Group (not applicable to SIP-T20P)
ACD
Hot Desking
Zero Touch
URL (not applicable to SIP-T20P)
Keypad Lock
Directory
Integer
For the memory key and programable key, the default value is not applicable.
For the line key, when x=1, the default value is 1.
When x=2, the default value is 2.
…
When x=6, the default value is 6.
Valid values are:
1 to 6 (for SIP-T28P)
1 to 3 (for SIP-T26P/T22P)
1 to 2 (for SIP-T21P/T20P)
1 (for SIP-T19P)
1-Line 1
2-Line 2
…
6-Line 6 memorykey.1.line = 2
Configuration File
<y0000000000xx>.cfg
Configures the value for some key features.
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Format
Default Value
Range
Example
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
String
Blank
String within 99 characters
When you assign the Speed Dial to the memory key, this parameter is used to specify the number you want to dial out. memorykey.1.value = 1001
Parameter- memorykey.X.pickup_value
Parameter- linekey.X.pickup_value
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures the pickup code for BLF feature.
This parameter is only applicable to BLF feature.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
String
Blank
String within 256 characters memorykey.1.pickup_value = *88
Parameter- memorykey.X.xml_phonebook
Configuration File
<y0000000000xx>.cfg
Parameter- linekey.X.xml_phonebook
Parameter- programablekey.X.xml_phone
book
406
Description
Format
Default Value
Range
Example
Appendix
Configures the desired group or remote phone book when multiple groups or remote phone books are configured on the IP phone.
This parameter is only applicable to Local
Group/XML Group features.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
When the key feature is configured as Local
Group, valid values are:
0-All contacts
1-First local group
…
5-Fifth local group
When the key feature is configured as XML
Group (remote phone book), valid values are:
0-First XML group
1-Second XML group
…
4-Fifth XML group
Integer
0
0 to 5
Configures the second remote phone book. memorykey.1.xml_phonebook = 1
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Keypad Lock Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Parameter- programablekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Configures a DSS key as a keypad lock key on the IP phone.
The digit 50 stands for the key type Keypad
Lock.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
Integer
50 memorykey.1.type = 50
DND Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Parameter- programablekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Configures a DSS key as a DND key on the IP phone.
The digit 5 stands for the key type DND.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
408
Format
Value
Example
Appendix
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
Integer
5 memorykey.1.type = 5
Directed Call Pickup Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Parameter- programablekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Configures a DSS key as a directed call pickup key on the IP phone.
The digit 9 stands for the key type Directed
Pickup.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
Integer
9 memorykey.1.type = 9
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Parameter- memorykey.X.line
Parameter- linekey.X.line
Parameter- programablekey.X.line
Configuration File
<y0000000000xx>.cfg
Description
Format
Range
Configures the desired line to apply the directed call pickup key.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
Integer
Valid values are:
1 to 6 (for SIP-T28P)
1 to 3 (for SIP-T26P/T22P)
1 to 2 (for SIP-T21P/T20P)
1 (for SIP-T19P)
1-Line 1
2-Line 2
…
6-Line 6 memorykey.1.line = 1 Example
Parameter- memorykey.X.value
Parameter- linekey.X.value
Parameter- programablekey.X.value
Configuration File
<y0000000000xx>.cfg
Description
Configures the directed call pickup feature code followed by the monitored extension.
For the memory key, x ranges from 1 to 10.
410
Format
Range
Example
Group Call Pickup Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Parameter- programablekey.X.type
Appendix
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
String
String within 99 characters memorykey.1.value = *971001
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Configures a DSS key as a group call pickup key on the IP phone.
The digit 23 stands for the key type Group
Pickup.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
Integer
23 memorykey.1.type = 23
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Parameter- memorykey.X.line
Parameter- linekey.X.line
Parameter- programablekey.X.line
Configuration File
<y0000000000xx>.cfg
Description
Format
Range
Example
Configures the desired line to apply the group call pickup key.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
Integer
Valid values are:
1 to 6 (for SIP-T28P)
1 to 3 (for SIP-T26P/T22P)
1 to 2 (for SIP-T21P/T20P)
1 (for SIP-T19P)
1-Line 1
2-Line 2
…
6-Line 6 memorykey.1.line = 1
Parameter- memorykey.X.value
Parameter- linekey.X.value
Parameter- programablekey.X.value
Configuration File
<y0000000000xx>.cfg
Description
Configures the group call pickup feature code.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
412
Format
Range
Example
Call Return Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Parameter- programablekey.X.type
Appendix
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
String
String within 99 characters memorykey.1.value = *98
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Configures a DSS key as a call return key on the
IP phone.
The digit 7 stands for the key type Call Return.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14
(For SIP-T19P IP phones, x=1-9, 13, 14; For
SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
Integer
7 memorykey.1.type = 7
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Call Park Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Configures a DSS key as a call park key on the
IP phone.
The digit 10 stands for the key type Call Park.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
Integer
10 memorykey.1.type = 10
Parameter- memorykey.X.line
Parameter- linekey.X.line
Configuration File
<y0000000000xx>.cfg
Description
Format
Range
Example
Configures the desired line to apply the call park key.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
Integer
Valid values are:
1 to 6 (for SIP-T28P)
1 to 3 (for SIP-T26P/T22P)
1 to 2 (for SIP-T21P/T20P)
1-Line 1
2-Line 2
…
6-Line 6 memorykey.1.line = 1
414
Parameter- memorykey.X.value
Parameter- linekey.X.value
Description
Format
Range
Example
Intercom Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Description
Format
Value
Example
Parameter- memorykey.X.line
Parameter- linekey.X.line
Description
Appendix
Configuration File
<y0000000000xx>.cfg
Configures the call park feature code.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
String
String within 99 characters memorykey.1.value = *99
Configuration File
<y0000000000xx>.cfg
Configures a DSS key as an intercom key.
The digit 14 stands for the key type Intercom.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
Integer
14 memorykey.1.type = 14
Configuration File
<y0000000000xx>.cfg
Configures the desired line to apply the intercom key.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
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Format
Range
Example
Note: It is not applicable to SIP-T19P IP phones.
Integer
Valid values are:
1 to 6 (for SIP-T28P)
1 to 3 (for SIP-T26P/T22P)
1 to 2 (for SIP-T21P/T20P)
1-Line 1
2-Line 2
…
6-Line 6 memorykey.1.line = 1
Parameter- memorykey.X.value
Parameter- linekey.X.value
Configuration File
<y0000000000xx>.cfg
Description
Format
Range
Example
Configures the intercom number.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
String
String within 99 characters memorykey.1.value = 1008
LDAP Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Parameter- programablekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Configures a DSS key as an LDAP key on the IP phone.
The digit 38 stands for the key type LDAP.
For the memory key, x ranges from 1 to 10.
416
Format
Value
Example
BLF Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Description
Format
Value
Example
Parameter- memorykey.X.line
Parameter- linekey.X.line
Description
Format
Appendix
For the line key, x ranges from 1 to 6.
For the programable key, x ranges from 1 to 14.
(For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P
IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
Note: It is not applicable to SIP-T19P and SIP-T20P
IP phones.
Integer
38 memorykey.1.type = 38
Configuration File
<y0000000000xx>.cfg
Configures a DSS key as a BLF key on the IP phone.
The digit 16 stands for the key type BLF.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
Integer
16 memorykey.1.type = 16
Configuration File
<y0000000000xx>.cfg
Configures the desired line to apply the BLF key.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
Integer
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Range
Example
Valid values are:
1 to 6 (for SIP-T28P)
1 to 3 (for SIP-T26P/T22P)
1 to 2 (for SIP-T21P/T20P)
1-Line 1
2-Line 2
…
6-Line 6 memorykey.1.line = 1
Parameter- memorykey.X.value
Parameter- linekey.X.value
Configuration File
<y0000000000xx>.cfg
Description
Format
Range
Example
Configures the number of the monitored user.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
String
String within 99 characters memorykey.1.value = 1008
Parameter- memorykey.X.pickup_value
Parameter- linekey.X.pickup_value
Configuration File
<y0000000000xx>.cfg
Description
Format
Default Value
Range
Example
Configures the pickup code for BLF feature.
This parameter only applies to BLF feature.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
String
Blank
String within 256 characters memorykey.1.pickup_value = *88
418
Appendix
ACD Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Multicast Paging Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Configures a DSS key as an ACD key on the IP phone.
The digit 42 stands for the key type ACD.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
Integer
42 memorykey.1.type = 42
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Configures a DSS key as a multicast paging key on the IP phone.
The digit 24 stands for the key type Multicast
Paging.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
Integer
24 memorykey.1.type = 24
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Parameter- memorykey.X.value
Parameter- linekey.X.value
Configuration File
<y0000000000xx>.cfg
Description
Format
Range
Example
Record Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Configures the multicast IP address and port number.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
The valid multicast IP addresses range from
224.0.0.0 to 239.255.255.255.
IP Address
224.0.0.0 to 239.255.255.255 memorykey.1.value = 224.5.5.6:10008
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Configures a DSS key as a record key on the IP phone.
The digit 25 stands for the key type Record.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
Integer
25 memorykey.1.type = 25
420
Appendix
URL Record Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Parameter- memorykey.X.value
Parameter- linekey.X.value
Description
Format
Default Value
Range
Example
Configures a DSS key as a URL record key on the IP phone.
The digit 35 stands for the key type URL Record.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
Integer
35 memorykey.1.type = 35
Configuration File
<y0000000000xx>.cfg
Configures the URL to record a call.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
Note: It is not applicable to SIP-T19P IP phones.
String
Blank
String within 99 characters memorykey.1.value = http://10.1.2.224/phonerecording.cgi
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Hot Desking Key
Parameter- memorykey.X.type
Parameter- linekey.X.type
Parameter- programablekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Format
Value
Example
Configures a DSS key as a hot desking key on the IP phone.
The digit 34 stands for the key type Hot Desking.
For the memory key, x ranges from 1 to 10.
For the line key, x ranges from 1 to 6.
For the programable key, x=1-9, 13, 14.
Note: You can configure a programable key as a hot desking key on SIP-T19P IP phones only.
Integer
34 memorykey.1.type = 34
This section describes how Yealink IP phones comply with the IETF definition of SIP as described in RFC 3261.
This section contains compliance information in the following:
RFC and Internet Draft Support
SIP Session Description Protocol (SDP) Usage
422
The following RFC’s and Internet drafts are supported:
RFC 1321—The MD5 Message-Digest Algorithm
RFC 1889—RTP Media control
Appendix
RFC 2112—Multipart MIME
RFC 2246—The TLS Protocol Version 1.0
RFC 2327—SDP: Session Description Protocol
RFC 2543—SIP: Session Initiation Protocol
RFC 2616—Hypertext Transfer Protocol -- HTTP/1.1
RFC 2617—Http Authentication: Basic and Digest access authentication
RFC 2782—A DNS RR for specifying the location of services (DNS SRV)
RFC 2806—URLs for Telephone Calls
RFC 2833—RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC2915—The Naming Authority Pointer (NAPTR) DNS Resource Record
RFC 3087—Control of Service Context using SIP Request-URI
RFC 3261—SIP: Session Initiation Protocol (replacement for RFC 2543)
RFC 3262—Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
RFC 3263—Session Initiation Protocol (SIP): Locating SIP Servers
RFC 3264—An Offer/Answer Model with the Session Description Protocol (SDP)
RFC 3265—Session Initiation Protocol (SIP) - Specific Event Notification
RFC 3266—Support for IPv6 in Session Description Protocol (SDP)
RFC 3310—HTTP Digest Authentication Using Authentication and Key Agreement
(AKA)
RFC 3311—The Session Initiation Protocol (SIP) UPDATE Method
RFC 3312—Integration of Resource Management and SIP
RFC 3313—Private SIP Extensions for Media Authorization
RFC 3323—A Privacy Mechanism for the Session Initiation Protocol (SIP)
RFC 3324—Requirements for Network Asserted Identity
RFC 3325—SIP Asserted Identity
RFC 3326—The Reason Header Field for the Session Initiation Protocol (SIP)
RFC 3361—DHCP-for-IPv4 Option for SIP Servers
RFC 3372—SIP for Telephones (SIP-T): Context and Architectures
RFC 3420—Internet Media Type message/sipfrag
RFC 3428—Session Initiation Protocol (SIP) Extension for Instant Messaging
RFC 3455—Private Header (P-Header) Extensions to the SIP for the 3GPP
RFC 3486—Compressing the Session Initiation Protocol (SIP)
RFC 3489—STUN - Simple Traversal of User Datagram Protocol (UDP) Through
Network Address Translators (NATs)
RFC 3515—The Session Initiation Protocol (SIP) Refer Method
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
RFC 3550—RTP , RTCP, IETF RFC 3550
RFC 3556—Session Description Protocol (SDP) Bandwidth Modifiers for RTCP
Bandwidth
RFC 3581—An Extension to the SIP for Symmetric Response Routing
RFC 3608—SIP Extension Header Field for Service Route Discovery During
Registration
RFC 3665—Session Initiation Protocol (SIP) Basic Call Flow Examples
RFC 3666—SIP Public Switched Telephone Network (PSTN) Call Flows.
RFC 3680—SIP Event Package for Registrations
RFC 3702—Authentication, Authorization, and Accounting Requirements for the SIP
RFC 3711—The Secure Real-time Transport Protocol (SRTP)
RFC 3725—Best Current Practices for Third Party Call Control (3pcc) in the Session
Initiation Protocol (SIP)
RFC 3842—A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)
RFC 3856—A Presence Event Package for Session Initiation Protocol (SIP)
RFC 3890—A Transport Independent Bandwidth Modifier for the SDP
RFC 3891—The Session Initiation Protocol (SIP) “Replaces” Header
RFC 3892—The Session Initiation Protocol (SIP) Referred-By Mechanism
RFC 3959—The Early Session Disposition Type for SIP
RFC 3960—Early Media and Ringing Tone Generation in SIP
RFC3966—The tel URI for telephone number
RFC 3968—The Internet Assigned Number Authority (IANA) Header Field
Parameter Registry for the Session Initiation Protocol (SIP)
RFC 3969—The Internet Assigned Number Authority (IANA) Uniform Resource
Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP)
RFC 4028—Session Timers in the Session Initiation Protocol (SIP)
RFC 4235—An INVITE-Initiated Dialog Event Package for the Session Initiation
Protocol (SIP)
RFC 4244—An Extension to the SIP for Request History Information
RFC 4317—Session Description Protocol (SDP) Offer/Answer Examples
RFC 4353—A Framework for Conferencing with the SIP
RFC 4475—Session Initiation Protocol (SIP) Torture
RFC 4485—Guidelines for Authors of Extensions to the SIP
RFC 4504—SIP Telephony Device Requirements and Configuration
RFC 4566—SDP: Session Description Protocol.
424
Appendix
RFC 4568—Session Description Protocol (SDP) Security Descriptions for Media
Streams
RFC 4575—A SIP Event Package for Conference State
RFC 4579—SIP Call Control - Conferencing for User Agents
RFC 4662—A SIP Event Notification Extension for Resource Lists
RFC 5009—P-Early-Media Header
RFC 5079—Rejecting Anonymous Requests in SIP
RFC 5359—Session Initiation Protocol Service Examples
RFC 5589—Session Initiation Protocol (SIP) Call Control - Transfer draft-levy-sip-diversion-04.txt—Diversion Indication in SIP draft-ietf-sip-cc-transfer-05.txt—SIP Call Control - Transfer draft-anil-sipping-bla-02.txt—Implementing Bridged Line Appearances (BLA) Using
Session Initiation Protocol (SIP) draft-ietf-sip-privacy-04.txt—SIP Extensions for Network-Asserted Caller Identity and Privacy within Trusted Networks draft-ietf-sipping-cc-conferencing-03.txt—SIP Call Control - Conferencing for User
Agents
To find the applicable Request for Comments (RFC) document, go to http://www.ietf.org/rfc.html
and enter the RFC number.
The following SIP request messages are supported:
Method
REGISTER
Supported
Yes
INVITE
ACK
CANCEL
BYE
OPTIONS
SUBSCRIBE
Yes
Yes
Yes
Yes
Yes
Yes
Notes
Yealink IP phones support mid-call changes such as placing a call on hold as signaled by a new INVITE that contains an existing
Call-ID.
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Method
NOTIFY
REFER
PRACK
INFO
MESSAGE
UPDATE
PUBLISH
Supported
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Notes
426
The following SIP request headers are supported:
Method
Accept
Alert-Info
Allow
Allow-Events
Authorization
Call-ID
Call-Info
Contact
Content-Length
Content-Type
CSeq
Diversion
Event
Expires
From
Max-Forwards
Min-SE
P-Asserted-Identity
Supported
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Notes
Method
P-Preferred-Identity
Proxy-Authenticate
Proxy-Authorization
RAck
Record-Route
Refer-To
Referred-By
Remote-Party-ID
Replaces
Require
Route
RSeq
Session-Expires
Subscription-State
Supported
To
User-Agent
Via
Supported
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Appendix
Notes
The following SIP responses are supported:
1xx Response—Information Responses
1xx Response
100 Trying
180 Ringing
181 Call Is Being Forwarded
183 Session Progress
Supported
Yes
Yes
Yes
Yes
2xx Response—Successful Responses
Notes
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
2xx Response
200 OK
202 Accepted
Supported
Yes
Yes
3xx Response—Redirection Responses
3xx Response
300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
Supported
Yes
Yes
Yes
4xx Response—Request Failure Responses
4xx Response
400 Bad Request
401 Unauthorized
402 Payment Required
403 Forbidden
404 Not Found
405 Method Not Allowed
406 Not Acceptable
407 Proxy Authentication
Required
408 Request Timeout
409 Conflict
410 Gone
411 Length Required
413 Request Entity Too Large
414 Request-URI Too Long
415 Unsupported Media Type
416 Unsupported URI Scheme
420 Bad Extension
421 Extension Required
Supported
Yes
Yes
Yes
Yes
Yes
Yes
No
Yes
Yes
No
No
No
No
Yes
Yes
No
No
No
Notes
In REFER transfer.
Notes
Notes
428
4xx Response
423 Interval Too Brief
480 Temporarily Unavailable
481 Call/Transaction Does Not
Exist
482 Loop Detected
483 Too Many Hops
484 Address Incomplete
485 Ambiguous
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
491 Request Pending
493 Undecipherable
5xx Response—Server Failure Responses
Supported
Yes
Yes
Yes
No
Yes
Yes
Yes
Yes
No
Yes
No
No
5xx Response
500 Internal Server Error
501 Not Implemented
502 Bad Gateway
503 Service Unavailable
504 Gateway Timeout
505 Version Not Supported
Supported
Yes
Yes
No
No
No
No
6xx Response—Global Responses
6xx Response
600 Busy Everywhere
603 Decline
604 Does Not Exist Anywhere
606 Not Acceptable
Supported
Yes
Yes
No
No
Appendix
Notes
Notes
Notes
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
SDP Headers v—Protocol version o—Owner/creator and session identifier a—Media attribute c—Connection information m—Media name and transport address s—Session name t—Active time
Supported
Yes
Yes
Yes
Yes
Yes
Yes
Yes
SIP uses six request methods:
INVITE—Indicates a user is being invited to participate in a call session.
ACK—Confirms that the client has received a final response to an INVITE request.
BYE—Terminates a call and can be sent by either the caller or the callee.
CANCEL—Cancels any pending searches but does not terminate a call that has already been accepted.
OPTIONS—Queries the capabilities of servers.
REGISTER—Registers the address listed in the To header field with a SIP server.
The following types of responses are used by SIP and generated by the IP phone or the
SIP server:
SIP 1xx—Informational Responses
SIP 2xx—Successful Responses
SIP 3xx—Redirection Responses
SIP 4xx—Client Failure Responses
SIP 5xx—Server Failure Responses
SIP 6xx—Global Failure Responses
430
Appendix
The following figure illustrates the scenario of a successful call. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones.
The call flow scenario is as follows:
1. User A calls User B.
2. User B answers the call.
3. User B hangs up.
User A Proxy Server User B
Step
F1
F1. INVITE B
F2. INVITE B
F3. 100 Trying
F4. 100 Trying
F5. 180 Ringing
F6. 180 Ringing
F7. 200 OK
F8. 200 OK
F9. ACK
F10. ACK
2-way RTP channel established
F11. BYE
F12. BYE
F13. 200 OK
Action
INVITE—User A to Proxy
Server
F14. 200 OK
Description
User A sends a SIP INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted
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Step
F2
F3
F4
F5
F6
F7
F8
INVITE—Proxy Server to User
B
100 Trying—User B to Proxy
Server
100 Trying—Proxy Server to
User A
180 Ringing—User B to Proxy
Server
180 Ringing—Proxy Server to
User A
200 OK— User B to Proxy
Server
Action
200OK—Proxy Server to User
A
Description in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User B. The proxy server sends the INVITE message to User B.
User B sends a SIP 100 Trying response to the proxy server. The 100 Trying response indicates that the INVITE request has been received by User B.
The proxy server forwards the SIP 100
Trying to User A to indicate that the
INVITE request has been received by
User B.
User B sends a SIP 180 Ringing response to the proxy server. The 180 Ringing response indicates that the User B is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User B is being alerted.
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the connection has been made.
The proxy server forwards the 200 OK message to User A. The 200 OK
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Appendix
Step
F9
F10
F11
F12
F13
F14
Action
ACK—User A to Proxy Server
ACK—Proxy Server to User B
Description response notifies User A that the connection has been made.
User A sends a SIP ACK to the proxy server. The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server sends the SIP ACK to
User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
BYE—User B to Proxy Server
BYE—Proxy Server to User A
200 OK—User A to Proxy
Server
200 OK—Proxy Server to User
B
User B terminates the call session by sending a SIP BYE request to the proxy server. The BYE request indicates that
User B wants to release the call.
The proxy server forwards the SIP BYE request to User A to notify that User B wants to release the call.
User A sends a SIP 200 OK response to the proxy server. The 200 OK response indicates that User A has received the
BYE request. The call session is now terminated.
The proxy server forwards the SIP 200
OK response to User B to indicate that
User A has received the BYE request.
The call session is now terminated.
The following figure illustrates the scenario of an unsuccessful call caused by the called user’s being busy. In this scenario, the two end users are User A and User B. User A and
User B are located at Yealink SIP IP phones.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
The call flow scenario is as follows:
1. User A calls User B.
2. User B is busy on the IP phone and unable or unwilling to take another call.
The call cannot be set up successfully.
434
User A Proxy Server User B
F1. INVITE B
F4. 100 Trying
F6. 486 Busy Here
F7. ACK
F2. INVITE B
F3. 100 Trying
F5. 486 Busy Here
Step Action
F1
INVITE—User A to Proxy
Server
F8. ACK
Description
User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is
Appendix
Step
F2
F3
F4
F5
F6
F7
F8
Action
INVITE—Proxy Server to User
B
100 Trying—User B to Proxy
Server
100 Trying—Proxy Server to
User A
486 Busy Here—User B to
Proxy Server
486 Busy Here—Proxy Server to User A
ACK—User A to Proxy Server
ACK—Proxy Server to User B specified.
Description
The proxy server maps the SIP URI in the
To field to User B. Proxy server forwards the INVITE message to User B.
User B sends a SIP 100 Trying response to the proxy server. The 100 Trying response indicates that the INVITE request has been received by User B.
The proxy server forwards the SIP 100
Trying to User A to indicate that the
INVITE request has already been received.
User B sends a SIP 486 Busy Here response to the proxy server. The 486
Busy Here response is a client error response indicating that User B is successfully connected but User B is busy on the IP phone and unable or unwilling to take the call.
The proxy server forwards the 486 Busy
Here response to notify User A that User
B is busy.
User A sends a SIP ACK to the proxy server. The SIP ACK message indicates that User A has received the 486 Busy
Here message.
The proxy server forwards the SIP ACK to User B to indicate that the 486 Busy
Here message has already been received.
The following figure illustrates the scenario of an unsuccessful call caused by the called user’s no answering. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
The call flow scenario is as follows:
1. User A calls User B.
2. User B does not answer the call.
3. User A hangs up.
The call cannot be set up successfully.
User A
F1. INVITE B
F4. 180 Ringing
F5. CANCEL
F8. 200 OK
Proxy Server
F2. INVITE B
F3. 180 Ringing
F6. CANCEL
F7. 200 OK
User B
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Appendix
Step
F1
F2
F3
F4
F5
F6
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
B
180 Ringing—User B to Proxy
Server
CANCEL—User A to Proxy
Server
Action
180 Ringing—Proxy Server to
User A
CANCEL—Proxy Server to
Description
User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User B. Proxy server forwards the INVITE message to User B.
User B sends a SIP 180 Ringing response to the proxy server. The 180 Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User B is being alerted.
User A sends a SIP CANCEL request to the proxy server after not receiving an appropriate response within the time allocated in the INVITE request. The SIP
CANCEL request indicates that User A wants to disconnect the call.
The proxy server forwards the SIP
CANCEL request to notify User B that
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Step
F7
F8
User B
Action Description
User A wants to disconnect the call.
200 OK—User B to Proxy
Server
200 OK—Proxy Server to User
A
User B sends a SIP 200 OK response to the proxy server. The SIP 200 OK response indicates that User B has received the CANCEL request.
The proxy server forwards the SIP 200
OK response to notify User A that the
CANCEL request has been processed successfully.
The following figure illustrates a successful call setup and call hold. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones.
The call flow scenario is as follows:
1. User A calls User B.
2. User B answers the call.
438
Appendix
3. User A places User B on hold.
User A Proxy Server User B
F1. INVITE B
F4. 180 Ringing
F2. INVITE B
F3. 180 Ringing
F5. 200 OK
F6. 200 OK
F7. ACK
F8. ACK
2-way RTP channel established
F9. INVITE B (sendonly)
F10. INVITE B (sendonly)
F11. 200 OK
F12. 200 OK
F13. ACK
No RTP packets being sent
F14. ACK
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Step
F1
F2
F3
F4
F5
F6
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
B
Action Description
User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User B. The proxy server sends the INVITE message to User B.
180 Ringing—User B to Proxy
Server
180 Ringing—Proxy Server to
User A
200 OK—User B to Proxy
Server
User B sends a SIP 180 Ringing response to the proxy server. The 180 Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User B is being alerted.
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies the proxy server that the connection has been made.
200 OK—Proxy Server to User
A
The proxy server forwards the 200 OK message to User A. The 200 OK response notifies User A that the connection has been made.
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Appendix
Step
F7
F8
F9
F10
F11
F12
F13
F14
INVITE—User A to Proxy
Server
Action
ACK—User A to Proxy Server
ACK—Proxy Server to User B
Description
User A sends a SIP ACK to the proxy server. The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server sends the SIP ACK to
User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
User A sends a mid-call INVITE request to the proxy server with new SDP session parameters, which are used to place the call on hold.
The proxy server forwards the mid-call
INVITE message to User B.
INVITE—Proxy Server to User
B
200 OK—User B to Proxy
Server
200 OK—Proxy Server to User
A
ACK—User A to Proxy Server
ACK—Proxy Server to User B
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the INVITE is successfully processed.
The proxy server forwards the 200 OK response to User A. The 200 OK response notifies User B is successfully placed on hold.
User A sends an ACK message to the proxy server. The ACK confirms that
User A has received the 200 OK response. The call session is now temporarily inactive. No RTP packets are being sent.
The proxy server sends the ACK message to User B. The ACK confirms that the proxy server has received the
200 OK response.
The following figure illustrates a successful call between Yealink SIP IP phones in which two parties are in a call, one of the participants receives and answers an incoming call from a third party. In this call flow scenario, the end users are User A, User B, and User C.
They are all using Yealink SIP IP phones, which are connected via an IP network.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
The call flow scenario is as follows:
1. User A calls User B.
2. User B answers the call.
3. User C calls User B.
4. User B accepts the call from User C.
User A Proxy Server
F1. INVITE B
F2. INVITE B
F4. 180 Ringing
F3. 180 Ringing
F5. 200 OK
F6. 200 OK
F7. ACK
F8. ACK
2-way RTP channel established
User B
F10. INVITE A
F11. 180 Ringing
F13. INVITE B ( sendonly )
F14. INVITE B ( sendonly )
F15. 200 OK
F316 200 OK
F17. ACK
F18. ACK
No RTP Packets being sent
F19. 200 OK
F9. INVITE A
F12. 180 Ringing
User C
F20. 200 OK
F21. ACK
F22. ACK
2-way RTP channel established
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Appendix
Step
F1
F2
F3
F4
F5
F6
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
B
Action Description
User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User B. The proxy server sends the INVITE message to User B.
180 Ringing—User B to Proxy
Server
180 Ringing—Proxy Server to
User A
200 OK—User B to Proxy
Server
User B sends a SIP 180 Ringing response to the proxy server. The 180 Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User B is being alerted.
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies proxy server that the connection has been made.
200 OK—Proxy Server to User
A
The proxy server forwards the 200 OK message to User A. The 200 OK response notifies User A that the connection has been made.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Step
F7
F8
F9
F10
F11
F12
INVITE—User C to Proxy
Server
INVITE—Proxy Server to User
A
Action
ACK—User A to Proxy Server
ACK—Proxy Server to User B
180 Ringing—User A to Proxy
Server
180 Ringing—Proxy Server to
User C
Description
User A sends a SIP ACK to the proxy server, The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server sends the SIP ACK to
User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
User C sends a SIP INVITE message to the proxy server. The INVITE request is an invitation to User A to participate in a call session.
In the INVITE request:
The IP address of User A is inserted in the Request-URI field.
User C is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User C is ready to receive is specified.
The port on which User A is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User A. The proxy server sends the INVITE message to User A.
User A sends a SIP 180 Ringing response to the proxy server. The 180
Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User C. User C hears the ring-back tone indicating that
User A is being alerted.
444
Appendix
Step
F13
F14
F15
F16
F17
F18
F19
F20
F21
F22
INVITE—User A to Proxy
Server
Action Description
User A sends a mid-call INVITE request to the proxy server with new SDP session parameters, which are used to place the call on hold.
INVITE—Proxy Server to User
B
200 OK—User B to Proxy
Server
200 OK—Proxy Server to User
A
ACK—User A to Proxy Server
ACK—Proxy Server to User B
The proxy server forwards the mid-call
INVITE message to User B.
User B sends a 200 OK to the proxy server. The 200 OK response indicates that the INVITE was successfully processed.
The proxy server forwards the 200 OK response to User A. The 200 OK response notifies User B is successfully placed on hold.
User A sends an ACK message to the proxy server. The ACK confirms that
User A has received the 200 OK response. The call session is now temporarily inactive. No RTP packets are being sent.
The proxy server sends the ACK message to User B. The ACK confirms that the proxy server has received the
200 OK response.
200 OK—User A to Proxy
Server
200 OK—Proxy Server User C
ACK—User C to Proxy Server
ACK—Proxy Server to User A
User A sends a 200 OK response to the proxy server. The 200 OK response notifies that the connection has been made.
The proxy server forwards the 200 OK message to User C.
User C sends a SIP ACK to the proxy server. The ACK confirms that User C has received the 200 OK response. The call session is now active.
The proxy server forwards the SIP ACK to User A to confirm that User C has received the 200 OK response.
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
The following figure illustrates a successful call between Yealink SIP IP phones in which two parties are in a call and then one of the parties transfers the call to a third party without consultation. This is called a blind transfer. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
The call flow scenario is as follows:
1. User A calls User B.
2. User B answers the call.
3. User B transfers the call to User C.
446
4. User C answers the call.
Call is established between User A and User C.
Appendix
User A Proxy Server
F1. INVITE B
F2. INVITE B
F3. 180 Ringing
F4. 180 Ringing
F6. 200 OK
F7. ACK
F8. ACK
2-way RTP channel established
F9. REFER
F10. 202 Accepted
F11. REFER
F5. 200 OK
F12. 202 Accepted
F17. BYE
F18. BYE
F19. 200 OK
F20. 200 OK
F21. INVITE C
User B
F22. INVITE C
F23. 180 Ringing
F24. 180 Ringing
F26. 200 OK
F27. ACK
F25. 200 OK
F28. ACK
2-way RTP channel established
User C
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Step
F1
F2
F3
F4
F5
F6
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
B
Action Description
User A sends an INVITE message to the proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User B. The proxy server sends the INVITE message to User B.
180 Ringing—User B to Proxy server
180 Ringing—Proxy Server to
User A
200 OK—User B to Proxy
Server
User B sends a SIP 180 Ringing response to the proxy server. The 180 Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User B is being alerted.
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the connection has been made.
200 OK—Proxy Server to User
A
The proxy server forwards the 200 OK message to User A. The 200 OK response notifies User A that the connection has been made.
448
Appendix
Step
F7
F8
F9
F10
F11
F12
F13
F14
F15
F16
F17
Action
ACK—User A to Proxy Server
Description
User A sends a SIP ACK to the proxy server, The ACK confirms that User A has received the 200 OK response. The call session is now active.
ACK—Proxy Server to User B
202 Accepted—Proxy Server to User B
The proxy server sends the SIP ACK to
User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
REFER—User B to Proxy Server
User B sends a REFER message to the proxy server. User B performs a blind transfer of User A to User C.
The proxy server sends a SIP 202 Accept response to User B. The 202 Accepted response notifies User B that the proxy server has received the REFER message.
REFER—Proxy Server to User
A
202 Accepted—User A to
Proxy Server
BYE—User B to Proxy Server
The proxy server forwards the REFER message to User A.
User A sends a SIP 202 Accept response to the proxy server. The 202 Accepted response indicates that User A accepts the transfer.
User B terminates the call session by sending a SIP BYE request to the proxy server. The BYE request indicates that
User B wants to release the call.
BYE—Proxy Server to User A
200OK—User A to Proxy
Server
200OK—Proxy Server to User
B
INVITE—User A to Proxy
Server
The proxy server forwards the BYE request to User A.
User A sends a SIP 200 OK response to the proxy server. The 200 OK response confirms that User A has received the
BYE request.
The proxy server forwards the SIP 200
OK response to User B.
User A sends a SIP INVITE request to the proxy server. In the INVITE request, a unique Call-ID is generated and the
Contact-URI field indicates that User A
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Step
F18
F19
F20
F21
F22
F23
F24
INVITE—Proxy Server to User
C
180 Ringing—User C to Proxy
Server
180 Ringing—Proxy Server to
User A
200OK—User C to Proxy
Server
200OK—Proxy Server to User
A
Action
ACK— User A to Proxy Server
ACK—Proxy Server to User C
Description requests the call.
The proxy server maps the SIP URI in the
To field to User C.
User C sends a SIP 180 Ringing response to the proxy server. The 180
Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User C is being alerted
User C sends a SIP 200 OK response to the proxy server. The 200 OK response notifies the proxy server that the connection has been made.
The proxy server forwards the SIP 200
OK response to User A.
User A sends a SIP ACK to the proxy server. The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server forwards the ACK message to User C. The ACK confirms that User A has received the 200 OK response. The call session is now active.
450
The following figure illustrates a successful call between Yealink SIP IP phones in which two parties are in a call and then one of the parties transfers the call to the third party with consultation. This is called attended transfer. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
The call flow scenario is as follows:
1. User A calls User B.
2. User B answers the call.
3. User A calls User C.
4. User C answers the call.
5. User A transfers the call to User C.
Call is established between User B and User C.
Appendix
User A Proxy Server User B
F1. INVITE B
F2. INVITE B
F3. 180 Ringing
F4. 180 Ringing
F5. 200 OK
F6. 200 OK
F7. ACK
F8. ACK
2-way RTP channel established
F9. INVITE B （sendonly）
F10. INVITE B （sendonly）
F11. 200 OK
F12. 200 OK
F13. ACK
F14. ACK
F15. INVITE C
F18. 180 Ringing
F20. 200 OK
F21. ACK
F16. INVITE C
F17. 180 Ringing
F19. 200 OK
F22. ACK
2-way RTP channel established
F23. REFER
F24. 202 Accepted
F25. REFER
F26. 202 Accepted
F31. BYE
F32. BYE
F33. 200 OK
F34. 200 OK
2-way RTP channel established
User C
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Step
F1
F2
F3
F4
F5
F6
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
B
Action Description
User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User B. The proxy server sends the INVITE message to User B.
180 Ringing—User B to Proxy
Server
180 Ringing—Proxy Server to
User A
200 OK—User B to Proxy
Server
User B sends a SIP 180 Ringing response to the proxy server. The 180 Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User B is being alerted.
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the connection has been made.
200 OK—Proxy Server to User
A
The proxy server forwards the 200 OK message to User A. The 200 OK response notifies User A that the connection has been made.
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Appendix
Step
F7
F8
F9
F10
F11
F12
F13
F14
F15
F16
INVITE—User A to Proxy
Server
Action
ACK—User A to Proxy Server
ACK—Proxy Server to User B
Description
User A sends a SIP ACK to the proxy server, The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server sends the SIP ACK to
User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
User A sends a mid-call INVITE request to the proxy server with new SDP session parameters, which are used to place the call on hold.
The proxy server forwards the mid-call
INVITE message to User B.
INVITE—Proxy Server to User
B
200 OK—User B to Proxy
Server
200 OK—Proxy Server to User
A
ACK—User A to Proxy Server
ACK—Proxy Server to User B
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the INVITE was successfully processed.
The proxy server forwards the 200 OK response to User A. The 200 OK response notifies User B is successfully placed on hold.
User A sends an ACK message to the proxy server. The ACK confirms that
User A has received the 200 OK response. The call session is now temporarily inactive. No RTP packets are being sent.
The proxy server sends the ACK message to User B. The ACK confirms that the proxy server has received the
200 OK response.
User A sends a SIP INVITE request to the proxy server. In the INVITE request, a unique Call-ID is generated and the
Contact-URI field indicates that User A requests the call.
The proxy server maps the SIP URI in the
To field to User C. The proxy server
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Step
F17
F18
F19
F20
F21
F22
F23
F24
F25
F26
C
180 Ringing—User C to Proxy
Server
180 Ringing—Proxy Server to
User A
200OK—User C to Proxy
Server
Action Description sends the INVITE request to User C.
User C sends a SIP 180 Ringing response to the proxy server. The 180
Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User C is being alerted.
User C sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the connection has been made.
200OK—Proxy Server to User
A
ACK— User A to Proxy Server
ACK—Proxy Server to User C
REFER—User A to Proxy
Server
202 Accepted—Proxy Server to User A
The proxy server forwards the SIP 200
OK response to User A. The 200 OK response notifies User A that the connection has been made.
User A sends a SIP ACK to the proxy server. The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server forwards the ACK message to User C. The ACK confirms that the proxy server has received the
200 OK response. The call session is now active.
User A sends a REFER message to the proxy server. User A performs a transfer of User B to User C.
The proxy server sends a SIP 202
Accepted response to User A. The 202
Accepted response notifies User A that the proxy server has received the REFER message.
REFER—Proxy Server to User B
The proxy server forwards the REFER message to User B.
202 Accepted—User B to
Proxy Server
User B sends a SIP 202 Accept response to the proxy server. The 202 Accepted
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Appendix
Step
F27
F28
F29
F30
200OK—User B to Proxy
Server
Action
BYE—User A to Proxy Server
BYE—Proxy Server to User B
200OK—Proxy Server to User
A
Description response indicates that User B accepts the transfer.
User A terminates the call session by sending a SIP BYE request to the proxy server. The BYE request indicates that
User A wants to release the call.
The proxy server forwards the BYE request to User B.
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that User B has received the BYE request.
The proxy server forwards the SIP 200
OK response to User A.
The following figure illustrates successful call forwarding between Yealink SIP IP phones in which User B has enabled always call forward. The incoming call is immediately forwarded to User C when User A calls User B. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
The call flow scenario is as follows:
1. User B enables always call forward, and the destination number is User C.
2. User A calls User B.
3. User B forwards the incoming call to User C.
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4. User C answers the call.
Call is established between User A and User C.
User A
F1. INVITE B
Proxy Server User B
F2. INVITE B
F3. 302 Move Temporarily
F4. ACK
F5. 302 Move Temporarily
F6. ACK
F7. INVITE C
F10. 180 Ringing
F12. 200 OK
F13. ACK
F8. INVITE C
F9. 180 Ringing
F11. 200 OK
F14. ACK
2-way RTP channel established
User C
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Appendix
Step
F1
F2
F3
F4
F5
F6
INVITE—User A to Proxy
Server
Action Description
User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of the User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
INVITE—Proxy Server to User
B
302 Move Temporarily—User
B to Proxy Server
ACK—Proxy Server to User B
The proxy server maps the SIP URI in the
To field to User B. The proxy server sends the INVITE message to User B.
User B sends a SIP 302 Moved
Temporarily message to the proxy server. The message indicates that User
B is not available at SIP phone B. User B rewrites the contact-URI.
The proxy server sends a SIP ACK to
User B, the ACK message notifies User B that the proxy server has received the
302 Move Temporarily message.
302 Move Temporarily—Proxy
Server to User A
The proxy server forwards the 302
Moved Temporarily message to User A.
ACK—User A to Proxy Server
User A sends a SIP ACK to the proxy server. The ACK message notifies the proxy server that User A has received the 302 Move Temporarily message.
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Step
F7
F8
F9
F10
F11
F12
F13
F14
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
C
180 Ringing—Proxy Server to
User A
200OK—User C to Proxy
Server
Action
180 Ringing—User C to Proxy
Server
200OK—Proxy Server to User
A
ACK—User A to Proxy Server
ACK—Proxy Server to User C
Description
User A sends a SIP INVITE request to the proxy server. In the INVITE request, a unique Call-ID is generated and the
Contact-URI field indicates that User A requested the call.
The proxy server maps the SIP URI in the
To field to User C. The proxy server sends the SIP INVITE request to User C.
User C sends a SIP 180 Ringing response to the proxy server. The 180
Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User C is being alerted.
User C sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the connection has been made.
The proxy server forwards the SIP 200
OK response to User A. The 200 OK response notifies User A that the connection has been made.
User A sends a SIP ACK to the proxy server. The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server forwards the ACK message to User C. The ACK confirms that the proxy server has received the
200 OK response. The call session is now active.
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The following figure illustrates successful call forwarding between Yealink SIP IP phones in which User B has enabled busy call forward. The incoming call is forwarded to User C when User B is busy. In this call flow scenario, the end users are User A, User B, and User
C. They are all using Yealink SIP IP phones, which are connected via an IP network.
The call flow scenario is as follows:
1. User B enables busy call forward, and the destination number is User C.
2. User A calls User B.
3. User B is busy.
4. User B forwards the incoming call to User C.
5. User C answers the call.
Call is established between User A and User C.
User C User A
F1. INVITE B
F4. 180 Ringing
Proxy Server User B
F2. INVITE B
F3. 180 Ringing
F5. 302 Move Temporarily
F6. ACK
F7. 302 Move Temporarily
F8. ACK
F9. INVITE C
F12. 180 Ringing
F14. 200 OK
F15. ACK
F10. INVITE C
F11. 180 Ringing
F13. 200 OK
F16. ACK
2-way RTP channel established
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Step
F1
F2
F3
F4
F5
F6
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
B
180 Ringing—User B to Proxy
Server
Action
180 Ringing—Proxy Server to
User A
302 Move Temporarily—User
B to Proxy Server
ACK—Proxy Server to User B
Description
User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User B. The proxy server sends the INVITE message to User B.
User B sends a SIP 180 Ringing response to the proxy server. The 180 Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User B is being alerted.
User B sends a SIP 302 Moved
Temporarily message to the proxy server. The message indicates that User
B is not available at SIP phone B. User B rewrites the contact-URI.
The proxy server sends a SIP ACK to
User B, the ACK message notifies User B that the proxy server has received the
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Appendix
Step
F7
F8
F9
F10
F11
F12
F13
F14
F15
F16
Action Description
ACK message.
302 Move Temporarily—Proxy
Server to User A
The proxy server forwards the 302
Moved Temporarily message to User A.
ACK—User A to Proxy Server
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
C
180 Ringing—User C to Proxy
Server
User A sends a SIP ACK to the proxy server. The ACK message notifies the proxy server that User A has received the ACK message.
User A sends a SIP INVITE request to the proxy server. In the INVITE request, a unique Call-ID is generated and the
Contact-URI field indicates that User A requests the call.
The proxy server forwards the SIP
INVITE request to User C.
User C sends a SIP 180 Ringing response to the proxy server. The 180
Ringing response indicates that the user is being alerted.
180 Ringing—Proxy Server to
User A
200OK—User C to Proxy
Server
200OK—Proxy Server to User
A
ACK— User A to Proxy Server
ACK—Proxy Server to User C
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User C is being alerted.
User C sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the connection has been made.
The proxy server forwards the SIP 200
OK response to User A.
User A sends a SIP ACK to the proxy server. The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server sends the ACK message to User C.
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The following figure illustrates successful call forwarding between Yealink SIP IP phones in which User B has enabled no answer call forward. The incoming call is forwarded to
User C when User B does not answer the incoming call after a period of time. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink
SIP IP phones, which are connected via an IP network.
The call flow scenario is as follows:
1. User B enables no answer call forward, and the destination number is User C.
2. User A calls User B.
3. User B does not answer the incoming call.
4. User B forwards the incoming call to User C.
5. User C answers the call.
Call is established between User A and User C.
User C
462
User A
F1. INVITE B
Proxy Server
F2. INVITE B
F3. 180 Ringing
F4. 180 Ringing
User B
F5. 302 Move Temporarily
F6. ACK
F7. 302 Move Temporarily
F8. ACK
F9. INVITE C
F12. 180 Ringing
F14. 200 OK
F15. ACK
F10. INVITE C
F11. 180 Ringing
F13. 200 OK
F16. ACK
2-way RTP channel established
Appendix
Step
F1
F2
F3
F4
F5
F6
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
B
180 Ringing—User B to Proxy
Server
Action
180 Ringing—Proxy Server to
User A
302 Move Temporarily—User
B to Proxy Server
ACK—Proxy Server to User B
Description
User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User B. The proxy server sends the INVITE message to User B.
User B sends a SIP 180 Ringing response to the proxy server. The 180 Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User B is being alerted.
User B sends a SIP 302 Moved
Temporarily message to the proxy server. The message indicates that User
B is not available at SIP phone B. User B rewrites the contact-URI.
The proxy server sends a SIP ACK to
User B, the ACK message notifies User B that the proxy server has received the
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Step
F7
F8
F9
F10
F11
F12
F13
F14
F15
F16
Action Description
ACK message.
302 Move Temporarily—Proxy
Server to User A
The proxy server forwards the 302
Moved Temporarily message to User A.
ACK—User A to Proxy Server
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
C
180 Ringing—User C to Proxy
Server
User A sends a SIP ACK to the proxy server. The ACK message notifies the proxy server that User A has received the ACK message.
User A sends a SIP INVITE request to the proxy server. In the INVITE request, a unique Call-ID is generated and the
Contact-URI field indicates that User A requests the call.
The proxy server forwards the SIP
INVITE request to User C.
User C sends a SIP 180 Ringing response to the proxy server. The 180
Ringing response indicates that the user is being alerted.
180 Ringing—Proxy Server to
User A
200OK—User C to Proxy
Server
200OK—Proxy Server to User
A
ACK— User A to Proxy Server
ACK—Proxy Server to User C
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User C is being alerted.
User C sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the connection has been made.
The proxy server forwards the SIP 200
OK response to User A. The 200 OK response notifies User A that the connection has been made.
User A sends a SIP ACK to the proxy server. The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server sends the ACK message to User C. The ACK confirms that the proxy server has received the
200 OK response.
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Appendix
The following figure illustrates successful 3-way calling between Yealink IP phones in which User A mixes two RTP channels and therefore establishes a conference between
User B and User C. In this call flow scenario, the end users are User A, User B, and User
C. They are all using Yealink SIP IP phones, which are connected via an IP network.
The call flow scenario is as follows:
1. User A calls User B.
2. User B answers the call.
3. User A places User B on hold.
4. User A calls User C.
5. User C answers the call.
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6. User A mixes the RTP channels and establishes a conference between User B and
User C.
User A Proxy Server
F1. INVITE B
F4. 180 Ringing
F2. INVITE B
F3. 180 Ringing
F5. 200 OK
F6. 200 OK
F7. ACK
F8. ACK
Session1 established between User A and User B is active
User B
F9. INVITE(sendonly)
Initiate three party conference
F12. 200 OK
F10. INVITE (sendonly)
F11. 200 OK
F13. ACK
F14. ACK
Session 1 established between User A and User B is hold
F15. INVITE C
F16. INVITE C
F17. 180 Ringing
F18. 180 Ringing
F20. 200 OK
F21. ACK
F19. 200 OK
F22. ACK
Both calls are active, come into three-party conference
User C
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Appendix
Step
F1
F2
F3
F4
F5
F6
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
B
Action Description
User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session.
In the INVITE request:
The IP address of User B is inserted in the Request-URI field.
User A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
The transaction number within a single call leg is identified in the
CSeq field.
The media capability User A is ready to receive is specified.
The port on which User B is prepared to receive the RTP data is specified.
The proxy server maps the SIP URI in the
To field to User B. Proxy server forwards the INVITE message to User B.
180 Ringing—User B to Proxy
Server
180 Ringing—Proxy Server to
User A
200 OK—User B to Proxy
Server
User B sends a SIP 180 Ringing response to the proxy server. The 180 Ringing response indicates that the user is being alerted.
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User B is being alerted.
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the connection has been made.
200 OK—Proxy Server to User
A
The proxy server forwards the 200 OK message to User A. The 200 OK response notifies User A that the connection has been made.
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Step
F7
F8
F9
F10
F11
F12
F13
F14
F15
F16
INVITE—User A to Proxy
Server
Action
ACK—User A to Proxy Server
ACK—Proxy Server to User B
Description
User A sends a SIP ACK to the proxy server. The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server sends the SIP ACK to
User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
User A sends a mid-call INVITE request to the proxy server with new SDP session parameters, which are used to place the call on hold.
The proxy server forwards the mid-call
INVITE message to User B.
INVITE—Proxy Server to User
B
200 OK—User B to Proxy
Server
200 OK—Proxy Server to User
A
ACK—User A to Proxy Server
ACK—Proxy Server to User B
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
User B sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the INVITE is successfully processed.
The proxy server forwards the 200 OK response to User A. The 200 OK response notifies User A that User B is successfully placed on hold.
User A sends the ACK message to the proxy server. The ACK confirms that
User A has received the 200 OK response. The call session is now temporarily inactive. No RTP packets are being sent.
The proxy server sends the ACK message to User B. The ACK confirms that the proxy server has received the
200 OK response.
User A sends a SIP INVITE request to the proxy server. In the INVITE request, a unique Call-ID is generated and the
Contact-URI field indicates that User A requests the call.
The proxy server maps the SIP URI in the
To field to User C. The proxy server
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Appendix
Step
F17
F18
F19
F20
F21
F22
C
Action Description sends the SIP INVITE request to User C.
180 Ringing—User C to Proxy
Server
User C sends a SIP 180 Ringing response to the proxy server. The 180
Ringing response indicates that the user is being alerted.
180 Ringing—Proxy Server to
User A
The proxy server forwards the 180
Ringing response to User A. User A hears the ring-back tone indicating that
User C is being alerted.
200OK—User C to Proxy
Server
User C sends a SIP 200 OK response to the proxy server. The 200 OK response notifies User A that the connection has been made.
200OK—Proxy Server to User
A
ACK— User A to Proxy Server
ACK—Proxy Server to User C
The proxy server forwards the SIP 200
OK response to User A. The 200 OK response notifies User A that the connection has been made.
User A sends a SIP ACK to the proxy server. The ACK confirms that User A has received the 200 OK response. The call session is now active.
The proxy server sends the ACK message to User C. The ACK confirms that the proxy server has received the
200 OK response.
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470
This section provides the sample configuration file necessary to configure the IP phone.
Any line beginning with a pound sign (#) is considered to be a comment, unless the # is contained within double quotes. For Boolean fields, 0 = disabled, 1 = enabled.
This file contains sample configurations for the <y0000000000xx>.cfg or <MAC>.cfg file.
The parameters included here are examples only. Not all possible parameters are shown in the sample configuration file. You can configure or comment the values as required. The settings in the <y0000000000xx>.cfg file will be overridden by settings in the <MAC>.cfg file.
Sample Configuration File
#!version:1.0.0.1
#Note: This file header cannot be edited or deleted.
#
Network Settings
network.internet_port.type =
#Configure the WAN port type; 0-DHCP, 1-PPPoE, 2-Static IP Address.
#If the WAN port type is configured as DHCP, you do not need to set the
#following network parameters.
#If the WAN port type is configured as Static IP Address, configure the
#following parameters. network.internet_port.ip = network.internet_port.mask = network.internet_port.gateway = network.primary_dns= network.secondary_dns =
#If the WAN port type is configured as PPPoE, configure the following
#parameters. network.pppoe.user = network.pppoe.password =
#
Dial Plan Settings
dialplan.area_code.code = dialplan.area_code.min_len = dialplan.area_code.max_len = dialplan.area_code.line_id = dialplan.block_out.number.1 = dialplan.block_out.line_id.1 = dialplan.dialnow.rule.1 = dialplan.dialnow.line_id.1 =
Appendix phone_setting.dialnow_delay = dialplan.replace.prefix.1 = dialplan.replace.replace.1 = dialplan.replace.line_id.1 = dialplan.item.1 =
#
Time Settings
local_time.time_zone = local_time.time_zone_name = local_time.ntp_server1 = local_time.ntp_server2 = local_time.interval = local_time.dhcp_time =
#Use the following parameters to set the time and date manually. local_time.manual_time_enable = local_time.date_format = local_time.time_format =
#
Auto DST Settings
local_time.summer_time = local_time.dst_time_type = local_time.start_time = local_time.end_time = local_time.offset_time =
#
Phone Lock
phone_setting.lock = phone_setting.phone_lock.unlock_pin = phone_setting.phone_lock.lock_time_out =
#
Language
lang.wui = lang.gui =
#Call Waiting
call_waiting.enable = call_waiting.tone =
#Auto Redial
auto_redial.enable = auto_redial.interval = auto_redial.times =
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
#Call Hold
features.play_hold_tone.enable = features.play_hold_tone.delay = sip.rfc2543_hold =
#Hotline
features.hotline_number = features.hotline_delay =
#Web Server Type
wui.http_enable = network.port.http = wui.https_enable = network.port.https =
#DTMF Suppression
features.dtmf.hide = features.dtmf.hide_delay =
#
Call Forward
# In Phone Mode
features.fwd_mode = 0 forward.always.enable = forward.always.target = forward.always.on_code = forward.always.off_code = forward.busy.enable = forward.busy.target = forward.busy.on_code = forward.busy.off_code = forward.no_answer.enable = forward.no_answer.target = forward.no_answer.timeout = forward.no_answer.on_code = forward.no_answer.off_code =
#In Custom Mode
features.fwd_mode = 1 account.1.always_fwd.enable = account.1.always_fwd.target = account.1.always_fwd.on_code = account.1.busy_fwd.off_code = account.1.busy_fwd.enable = account.1.busy_fwd.target =
472
Appendix account.1.busy_fwd.on_code = account.1.busy_fwd.off_code = account.1.timeout_fwd.enable = account.1.timeout_fwd.target = account.1.timeout_fwd.timeout = account.1.timeout_fwd.on_code = account.1.timeout_fwd.off_code =
#
Call Transfer
transfer.semi_attend_tran_enable = transfer.blind_tran_on_hook_enable = transfer.on_hook_trans_enable = transfer.tran_others_after_conf_enable =
#
Call Conference
account.1.conf_type = account.1.conf_uri =
#
DTMF
account.1.dtmf.type = account.1.dtmf.dtmf_payload = account.1.dtmf.info_type =
#
Distinctive Ring Tones
account.1.alert_info_url_enable = distinctive_ring_tones.alert_info.1.text = distinctive_ring_tones.alert_info.1.ringer =
#
Tones
voice.tone.dial = voice.tone.ring = voice.tone.busy = voice.tone.congestion = voice.tone.callwaiting = voice.tone.dialrecall = voice.tone.info = voice.tone.stutter = voice.tone.message = voice.tone.autoanswer =
#
Remote Phone Book
features.remote_phonebook.enable = features.remote_phonebook.flash_time =
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Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
#
LDAP
ldap.enable = ldap.name_filter = ldap.number_filter = ldap.host = ldap.port = ldap.base = ldap.user = ldap.password = ldap.max_hits = ldap.name_attr = ldap.numb_attr = ldap.display_name = ldap.version = ldap.call_in_lookup = ldap.ldap_sort =
#
Action URL
action_url.setup_completed = action_url.registered = action_url.unregistered = action_url.register_failed = action_url.off_hook = action_url.on_hook = action_url.incoming_call = action_url.outgoing_call = action_url.call_established = action_url.dnd_on = action_url.dnd_off = action_url.always_fwd_on = action_url.always_fwd_off = action_url.busy_fwd_on = action_url.busy_fwd_off = action_url.no_answer_fwd_on = action_url.no_answer_fwd_off = action_url.transfer_call = action_url.blind_transfer_call = action_url.attended_transfer_call = action_url.hold = action_url.unhold = action_url.mute = action_url.unmute = action_url.missed_call = action_url.call_terminated =
474
action_url.busy_to_idle = action_url.idle_to_busy = action_url.ip_change = action_url.forward_incoming_call = action_url.reject_incoming_call = action_url.answer_new_incoming_call = action_url.transfer_finished = action_url.transfer_failed =
#Access URL of Resource Files
dialplan_dialnow.url = dialplan_replace_rule.url = local_contact.data.url = remote_phonebook.data.1.url = directory_setting.url = super_search.url =
Appendix
475
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
476
Index
Numeric
A
Acoustic Echo Cancellation
Analyzing the Configuration Files 246
Appendix C: Configuration Parameters 260
Appendix E: SIP Call Flows 430
Appendix F: Sample Configuration File 470
Area Code
Audio Codecs
Automatic Call Distribution 149
B
C
Call Recording
Calling Line Identification Presentation
Comfort Noise Generation
Connected Line Identification Presentation 121
Configuring Basic Network Parameters 21
Configuring Security Features
Connect the Network and Power 13
Contrast
D
Dual Headset
E
Encrypting Configuration Files 219
477
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones
Enabling the Watch Dog Feature 245
G
Getting Information from Status Indicators 246
N
Network Address Translation (NAT) 187
Network Conference
H
Headset Prior
Hot Desking
Hotline
P
Physical Features of IP Phones 4
Power Indicator LED
I
Initialization Process Overview
Intercom
IPv6 Support
J
K
Key Features of IP Phones
Q
R
Reading Icons
RFC and Internet Draft Support
L
Local Contact File
Logo Customization
M
Message Waiting Indicator
Multicast Paging
S
Semi-attended Transfer
Server Redundancy
SIP
SIP Components
SIP IP Phone Models
SIP Request
SIP Session Description Protocol Usage
Specifying the Language to Use 58
478
Index
SRTP
Suppress DTMF Display
Summary of Changes
T
Time and Date
Transfer on Conference Hang Up 106
Transport Layer Security (TLS)
Troubleshooting Methods
U
Use Outbound Proxy in Dialog 92
User Password
V
Voice Activity Detection
VPN
W
Web User Interface
479
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