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Copyright © 2014 YEALINK NETWORK TECHNOLOGY
Copyright © 2014 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.
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 SIP-T4X IP phone firmware contains third-party software under the GNU General Public License
(GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms and conditions of the license.
The original GPL license, source code of components licensed under GPL and used in Yealink products can be downloaded online: http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293 .
About This Guide
The guide is intended for administrators who need to properly configure, customize, manage, and troubleshoot the IP phone system rather than the end-users. It provides details on the functionality and configuration of SIP-T4X IP phones.
Many of the features described in this guide involve network settings, which could affect the IP phone’s performance in the network. So an understanding of the IP networking and prior knowledge of IP telephony concepts are necessary.
This guide covers SIP-T48G, T46G, T42G and T41P IP phones. The following related documents for SIP-T4X IP phones are available:
Quick Start Guides, which describe how to assemble IP phones and configure the most basic features available on IP phones.
User Guides, which describe basic and advanced features available on IP phones.
Auto Provisioning Deployment Guide, which describes how to provision IP phones using the configuration files.
Configuration Conversion Tool User Guide, which describes how to convert and encrypt the configuration files using the Configuration Conversion Tool.
<y0000000000xx>.cfg and <MAC>.cfg template configuration files.
IP Phones Deployment Guide for BroadSoft UC-One Environments, which describes how to configure the BroadSoft features on the BroadWorks web portal and IP phones.
For support or service, please contact your Yealink reseller or go to Yealink Technical
Support online: http://www.yealink.com/Support.aspx
.
The information detailed in this guide is applicable to the firmware version 73 or higher.
The firmware format is like x.x.x.x.rom. The second x from left should be greater than or equal to 73 (e.g., the firmware version of SIP-T46G IP phone: 28.73.0.40.rom). This administrator guide includes the following chapters:
Chapter 1, “ Product Overview ” describes SIP components and SIP IP phones.
Chapter 2, “ Getting Started ” describes how to install and connect IP phones and
configuration methods.
Chapter 3, “ Configuring Basic Features ” describes how to configure basic features
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Administrator’s Guide for SIP-T4X IP Phones
on IP phones.
Chapter 4, “ Configuring Advanced Features ” describes how to configure
advanced features on IP phones.
Chapter 5, “ Configuring Audio Features ” describes how to configure audio features
on IP phones.
Chapter 6, “ Configuring Security Features ” describes how to configure security
features on IP phones.
Chapter 8, “ Resource Files ” describes the resource files that can be downloaded
by IP phones.
Chapter 9, “ Troubleshooting ” describes how to troubleshoot IP phones and
provides some common troubleshooting solutions.
Chapter 10, “ Appendix ” provides the glossary, reference information about IP
phones compliant with RFC 3261, SIP call flows and sample configuration files.
This section describes the changes to this guide for each release and guide version.
The following section is new for this version:
on page
Major updates have occurred to the following sections:
Physical Features of SIP-T4X IP Phones on page 4
on page
on page
Distinctive Ring Tones on page 260
on page
Voice Quality Monitoring on page 426
Appendix C: Configuring DSS Key on page 514
on page
vi
The following sections are new for this version:
on page
on page
Input Method Customization on page 99
on page
Feature Key Synchronization on page 211
on page
Capturing the Current Screen of the Phone
on page
on page
Major updates have occurred to the following sections:
on page
on page
Configuring Basic Network Parameters
on page
on page
Anonymous Call Rejection on page 158
on page
on page
on page
on page
on page
on page
Transport Layer Security on page 461
Secure Real-Time Transport Protocol
on page
Encrypting Configuration Files on page 474
Analyzing Configuration Files on page 506
The following sections are new for this version:
Provisioning Server on page 18
Background Noise Suppression on page 454
About This Guide vii
Administrator’s Guide for SIP-T4X IP Phones
Automatic Gain Control on page 454
Major updates have occurred to the following section:
Configuration Methods on page 16
on page
on page
This version is updated to incorporate SIP-T48G IP phones as one of the T4X device models. The following sections are new for this version:
on page
Major updates have occurred to the following section:
on page
Connecting the IP Phones on page 9
on page
on page
Logo Customization on page 103
on page
on page
on page
on page
viii
The following sections are new for this version:
on page
Major updates have occurred to the following sections:
Transport Layer Security on page 461
Major updates have occurred to the following section:
on page
Major updates have occurred to the following sections:
on page
on page
Major updates have occurred to the following sections:
on page
on page
on page
on page
Major updates have occurred to the following sections:
About This Guide ix
Administrator’s Guide for SIP-T4X IP Phones
on page
Encrypting Configuration Files
on page
This version is updated to incorporate T41P as one of the T4X device models. The following section is new for this version:
Logo Customization on page 103
Major updates have occurred to the following sections:
Configuring Transmission Methods of the Internet Port and PC Port on page 35
on page
on page
on page
Transport Layer Security on page 461
Secure Real-Time Transport Protocol on page 471
x
This version is updated to incorporate T42G as one of the T4X device models. The following section is new for this version:
on page
Major updates have occurred to the following sections:
TR-069 Device Management on page 409
on page
Appendix C: Configuring DSS Key
on page
Table of Contents
About This Guide ...................................................................... v
Table of Contents .................................................................... xi
Product Overview ..................................................................... 1
Getting Started ......................................................................... 9
xi
Administrator’s Guide for SIP-T4X IP Phones
Configuring Basic Features .................................................... 47
xii
Table of Contents
Configuring Advanced Features...........................................259
xiii
Administrator’s Guide for SIP-T4X IP Phones
Configuring Audio Features ..................................................423
Configuring Security Features ...............................................461
Resource Files ........................................................................481
xiv
Table of Contents
Troubleshooting .....................................................................493
What is the difference between a remote phone book and a local phone book? ....... 503
What is the difference between user name, register name and display name? .......... 503
Why does the IP phone use DOB format logo file instead of popular BMP, JPG and so on?
Appendix ...............................................................................509
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Administrator’s Guide for SIP-T4X IP Phones
Index ......................................................................................569
xvi
Product Overview
This chapter contains the following information about SIP-T4X 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 the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries.
Session management provides the ability to control the attributes of an end-to-end call.
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Administrator’s Guide for SIP-T4X IP Phones
SIP provides capabilities to:
Determine the location of the target endpoint -- SIP supports address resolution, name mapping, and call redirection.
Determine the media capabilities of the target endpoint -- Via Session Description
Protocol (SDP), SIP determines the “lowest level” of common services between endpoints. Conferences are established using only the media capabilities that can be supported by all endpoints.
Determine the availability of the target endpoint -- A call cannot be completed because the target endpoint is unavailable. SIP determines whether the called party is already on the IP phone or does not answer in the allotted number of rings.
It then returns a message indicating why the target endpoint is unavailable.
Establish a session between the origin and target endpoint -- The call can be completed, SIP establishes a session between endpoints. SIP also supports mid-call changes, such as the addition of another endpoint to the conference or the change of a media characteristic or codec.
Handle the transfer and termination of calls -- SIP supports the transfer of calls from one endpoint to another. During a call transfer, SIP simply establishes a session between the transferee and a new endpoint (specified by the transferring party) and terminates the session between the transferee and the transferring party. At the end of a call, SIP terminates the sessions between all parties.
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 the following roles:
User Agent Client (UAC) -- A client application that initiates the SIP request.
User Agent Server (UAS) -- A server application that contacts the user when a SIP request is received and that returns a response on behalf of the user.
User Agent Client (UAC)
The UAC is an application that initiates up to six feasible SIP requests to the UAS. The six requests issued by the UAC are: INVITE, ACK, OPTIONS, BYE, CANCEL and REGISTER.
When the SIP session is being initiated by the UAC SIP component, the UAC determines the information essential for the request, which is the protocol, the port and the IP address of the UAS to which the request is being sent. This information can be dynamic and will make it challenging to put through a firewall. For this reason, it may be recommended to open the specific application type on the firewall. The UAC is also capable of using the information in the request URI to establish the course of the SIP request to its destination, as the request URI always specifies the host which is essential.
The port and protocol are not always specified by the request URI. Thus if the request does not specify a port or protocol, a default port or protocol is contacted. It may be
Product Overview preferential to use this method when not using an application layer firewall. Application layer firewalls like to know what applications are flowing though which ports and it is possible to use content types of other applications other than the one you are trying to let through what has been denied.
User agent server (UAS)
UAS is a server that hosts the application responsible for receiving the SIP requests from a UAC, and on reception it returns a response to the request back to the UAC. The UAS may issue multiple responses to the UAC, not necessarily a single response.
Communication between UAC and UAS is client/server and peer-to–peer.
Typically, a SIP endpoint is capable of functioning as both a UAC and a UAS, but it functions only as one or the other per transaction. Whether the endpoint functions as a
UAC or a UAS depends on the UA that initiates the request.
This section introduces the SIP-T4X IP phone family. SIP-T4X IP phones are endpoints in the overall network topology, which are designed to interoperate with other compatible equipments including application servers, media servers, internet-working gateways, voice bridges, and other endpoints. SIP-T4X IP phones are characterized by a large number of functions, which simplify business communication with a high standard of security and can work seamlessly with a large number of SIP PBXs.
SIP-T4X IP phones provide a powerful and flexible IP communication solution for Ethernet
TCP/IP networks, delivering excellent voice quality. The high-resolution graphic display provides content in multiple languages for system status, call log and directory access.
SIP-T4X IP phones also support advanced functionalities, including LDAP, Busy Lamp
Field, Sever Redundancy and Network Conference.
The following IP phone models are described:
SIP-T48G
SIP-T46G
SIP-T42G
SIP-T41P
SIP-T4X IP phones comply with the SIP standard (RFC 3261), and they can only be used within a network that supports this type of phone.
In order to operate as SIP endpoints in your network successfully, SIP-T4X IP phones must meet the following requirements:
A working IP network is established.
VoIP gateways are configured for SIP.
The latest (or compatible) firmware of SIP-T4X IP phones is available.
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Administrator’s Guide for SIP-T4X IP Phones
A call server is active and configured to receive and send SIP messages.
This section lists the available physical features of SIP-T4X IP phones.
SIP-T48G
4
Physical Features:
-
7” 800 x 480 pixel color touch screen with backlight
-
24 bit depth color
-
16 VoIP accounts, Broadsoft Validated/Asterisk
®
Compatible
-
HD Voice: HD Codec, HD Handset, HD Speaker
-
26 keys including 7 feature keys
-
1*RJ9 (4P4C) handset port
- 1*RJ9 (4P4C) headset port
- 2*RJ45 10/100/1000Mbps Ethernet ports
- 1*RJ12 (6P6C) expansion module port
- 4 LEDs: 1*power, 1*mute, 1*headset, 1*speakerphone
- Power adapter: AC 100~240V input and DC 5V/2A output
- Power over Ethernet (IEEE 802.3af)
- Built-in USB port, support Bluetooth headset
Product Overview
SIP-T46G
Physical Features:
-
4.3” 480 x 272 pixel color display with backlight
-
24 bit depth color
-
16 VoIP accounts, Broadsoft Validated/Asterisk ® Compatible
-
HD Voice: HD Codec, HD Handset, HD Speaker
-
40 keys including 10 line keys
-
1*RJ9 (4P4C) handset port
-
1*RJ9 (4P4C) headset port
- 2*RJ45 10/100/1000Mbps Ethernet ports
- 1*RJ12 (6P6C) expansion module port
-
14 LEDs: 1*power, 10*line, 1*mute, 1*headset, 1*speakerphone
- Power adapter: AC 100~240V input and DC 5V/2A output
- Power over Ethernet (IEEE 802.3af)
- Built-in USB port, support Bluetooth headset
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Administrator’s Guide for SIP-T4X IP Phones
SIP-T42G
Physical Features:
-
192 x 64 graphic LCD
-
12 VoIP accounts, Broadsoft Validated/Asterisk ® Compatible
-
HD Voice: HD Codec, HD Handset, HD Speaker
-
34 keys including 6 line keys
-
1*RJ9 (4P4C) handset port
-
1*RJ9 (4P4C) headset port
-
2*RJ45 10/100/1000Mbps Ethernet ports
- 1*RJ12 (6P6C) EHS36 headset adapter port
- 10 LEDs: 1*power, 6*line, 1*mute, 1*headset, 1*speakerphone
-
Power adapter: AC 100~240V input and DC 5V/1.2A output
- Power over Ethernet (IEEE 802.3af)
6
Product Overview
SIP-T41P
Physical Features:
-
192 x 64 graphic LCD
- 6 VoIP accounts, Broadsoft Validated/Asterisk ® Compatible
- HD Voice: HD Codec, HD Handset, HD Speaker
- 34 keys including 6 line keys
- 1*RJ9 (4P4C) handset port
- 1*RJ9 (4P4C) headset port
- 2*RJ45 10/100Mbps Ethernet ports
- 1*RJ12 (6P6C) EHS36 headset adapter port
- 10 LEDs: 1*power, 6*line, 1*mute, 1*headset, 1*speakerphone
- Power adapter: AC 100~240V input and DC 5V/1.2A output
- Power over Ethernet (IEEE 802.3af)
In addition to physical features introduced above, SIP-T4X IP phones also support the following key features when running the latest firmware:
Phone Features
-
Call Options: call waiting, call hold, call mute, call forward, call transfer, call pickup, conference.
- Basic Features: DND, auto redial, live dialpad, dial plan, hotline, caller
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Administrator’s Guide for SIP-T4X IP Phones
identity, auto answer.
-
Advanced Features: BLF, server redundancy, distinctive ring tones, remote phone book, LDAP, 802.1X authentication.
Codecs and Voice Features
-
Wideband codec: G.722
-
Narrowband codec: G.711 (A/μ), G.723, G.726, G.729, iLBC.
-
VAD, CNG, AEC, PLC, AJB, AGC
-
Full-duplex speakerphone with AEC
Network Features
-
SIP v1 (RFC2543), v2 (RFC3261)
-
IPv4/IPv6 support
-
NAT Traversal: STUN mode
-
DTMF: INBAND, RFC2833, SIP INFO
-
Proxy mode and peer-to-peer SIP link mode
-
IP assignment: Static/DHCP/PPPoE (PPPoE is for SIP-T48G/T46G only)
-
TFTP/DHCP client
- HTTP/HTTPS server
- DNS client
- NAT/DHCP server
Management
-
FTP/TFTP/HTTP/PnP auto-provision
-
Configuration: browser/phone/auto-provision
- Direct IP call without SIP proxy
- Dial number via SIP server
- Dial URL via SIP server
Security
-
HTTPS (server/client)
-
SRTP (RFC3711)
- Transport Layer Security (TLS)
- VLAN (802.1q), QoS
- Digest authentication using MD5/MD5-sess
- Secure configuration file via AES encryption
- Phone lock for personal privacy protection (not applicable to SIP-T48G)
- Admin/User configuration mode
8
Getting Started
This chapter provides basic information and installation instructions of SIP-T4X IP phones.
This chapter provides the following sections:
Initialization Process Overview
Configuring Basic Network Parameters
This section introduces how to install SIP-T4X IP phones with the components in packaging contents.
1. Attach the stand and optional wall mount bracket
2. Connect the handset and optional headset
3. Connect the network and power
Note
A headset, wall mount bracket and power adapter are not included in packaging contents.
1) Attach the stand and optional wall mount bracket:
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Administrator’s Guide for SIP-T4X IP Phones
For SIP-T46G:
For SIP-T48G/T42G/T41P:
Desk Mount Method
Desk Mount Method
Wall Mount Method (Optional)
Note
The top two slots on SIP-T48G IP phones are plugged up by silica gel. You need to pull out silica gel before attaching the wall mount bracket.
For more information on how to mount the IP phone to a wall, refer to Yealink Wall Mount
Quick Installation Guide for SIP-T4X IP Phones
.
10
2) Connect the handset, optional headset and Bluetooth headset:
For SIP-T48G/T46G:
Getting Started
For SIP-T42G/T41P:
Note
Wireless headset adapter EHS36 and Bluetooth USB dongle should be purchased separately.
For more information on how to use the EHS36 on the IP phone, refer to
Yealink EHS36
User Guide.
Bluetooth can only be used on the SIP-T48G and SIP-T46G IP phones. For more information on how to use the Bluetooth on SIP-T48G/T46G IP phones, refer to Yealink Bluetooth USB
Dongle BT40 User Guide
.
EXT port on SIP-T48G/T46G IP phones can also be used to connect the expansion module
EXP40. For more information on how to connect the EXP40, refer to
Yealink EXP40 User
Guide
.
3) Connect the network and power:
AC power
Power over Ethernet (PoE)
AC Power
To connect the AC power and network:
1. Connect the DC plug of the power adapter to the DC5V port on IP phones and connect the other end of the power adapter into an electrical power outlet.
11
Administrator’s Guide for SIP-T4X IP Phones
2. Connect the included or a standard Ethernet cable between the Internet port on IP phones and the one on the wall or switch/hub device port.
Power over Ethernet
With the included or a regular Ethernet cable, IP phones can be powered from a
PoE-compliant switch or hub.
To connect the PoE:
1. Connect the Ethernet cable between the Internet port on IP phones and an available port on the in-line power switch/hub.
Note
If in-line power switch/hub is provided, you don’t need to connect the IP phone to the power adapter. Make sure the switch/hub is PoE-compliant.
IP phones can also share the network with another network device such as a PC
(personal computer). It is an optional connection.
Important! Do not unplug or remove the power while IP phones are updating firmware and configurations.
12
The initialization process of IP phones is responsible for network connectivity and operation of IP phones in your local network.
Once you connect your IP phone to the network and to an electrical supply, the IP phone
Getting Started begins its initialization process.
During the initialization process, the following events proceed:
Loading the ROM file
The ROM file resides in the flash memory of IP phones. IP phones come from the factory with a ROM file preloaded. During initialization, IP phones run a bootstrap loader that loads and executes the ROM file.
Configuring the VLAN
If IP phones are connected to a switch, the switch notifies IP phones of the VLAN information defined on the switch (if using LLDP). IP phones can then proceed with the
DHCP request for its network settings (if using DHCP).
Querying the DHCP (Dynamic Host Configuration Protocol) Server
IP phones are capable of querying a DHCP server. DHCP is enabled on IP phones by default. The following network parameters can be obtained from the DHCP server during initialization:
IP Address
Subnet Mask
Gateway
Primary DNS (Domain Name Server)
Secondary DNS
You need to configure the network parameters of IP phones manually if any of them is not provided by the DHCP server. For more information on configuring network
parameters manually, refer to Configuring Network Parameters Manually on page 26 .
Contacting the auto provisioning server
SIP-T4X IP phones support the FTP, TFTP, HTTP, and HTTPS protocols for auto provisioning and are configured by default to use TFTP protocol. If IP phones are configured to obtain configurations from the TFTP server, they will connect to the TFTP server and download the configuration file(s) during startup. IP phones will be able to resolve and apply the configurations written in the configuration file(s). If IP phones do not obtain the configurations from the TFTP server, IP phones will use the configurations stored in the flash memory.
Updating firmware
If the access URL of the firmware is defined in the configuration file, the IP phone will download the firmware from the provisioning server. If the MD5 value of the downloaded firmware file differs from that of the image stored in the flash memory, the
IP phone will perform a firmware update.
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Administrator’s Guide for SIP-T4X IP Phones
Downloading the resource files
In addition to configuration file(s), IP phones may require resource files before it can deliver service. These resource files are optional, but if some particular features are being deployed, these files are required.
The followings show examples of resource files:
Language packs
Ring tones
Contact files
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 red.
2. The message “Initializing…Please wait” appears on the LCD screen when the IP phone starts up.
3. The main LCD screen displays the following:
Time and date
Soft key labels
4. Press the OK key to check the IP phone status, the LCD screen displays the valid IP address, MAC address, firmware version, etc.
If the IP phone has successfully passed through these steps, it starts up properly and is ready for use.
14
Icons associated with different features may appear on the LCD screen. The following table provides a description for each icon on SIP-T4X IP phone models.
T48G T46G T42G/T41P Description
Network is unavailable
Registered successfully
Register failed
Registering
(Flashing)
/
T48G
/
T46G
Getting Started
T42G/T41P Description
Hands-free speakerphone mode
Handset mode
/
Headset mode
Voice Mail
Text Message
Auto Answer
Do Not Disturb
Call Forward
Call Hold
Call Mute
Ringer volume is 0
Phone Lock
Multi-lingual lowercase letters input mode
Multi-lingual uppercase letters input mode
Alphanumeric input mode
Numeric input mode
Multi-lingual uppercase and lowercase letters input mode
Received Calls
Placed Calls
Missed Calls
Forwarded Calls
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Administrator’s Guide for SIP-T4X IP Phones
T48G T46G T42G/T41P Description
Recording box is full
/
A call cannot be recorded
Recording starts successfully
/
/
/
/
Recording cannot be started
Recording cannot be stopped
VPN is enabled
Bluetooth mode is on
Bluetooth headset is both paired and connected
Conference
The default contact icon
The default caller photo
IP phones can be configured automatically through configuration files stored on a central provisioning server, manually via the phone user interface or web user interface, or by a combination of the automatic and manual methods.
The recommended method for configuring IP phones is automatically through a central provisioning server. If a central provisioning server is not available, the manual method will allow changes to most features.
The following sections describe how to configure IP phones using each method.
16
An administrator or a user can configure and use IP phones via phone user interface.
Getting Started
Specific features access is restricted to the administrator. These specific features are password protected by default. The default password is “admin“(case-sensitive). Not all features are available on phone user interface. For more information, refer to Yealink phone-specific user guide, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
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). Most features are available for configuring via web user interface. IP phones support both HTTP and HTTPS protocols for accessing the web user
interface. For more information, refer to Web Server Type on page 65 .
An administrator can deploy and maintain a mass of IP phones using configuration files.
The configuration files consist of:
Common CFG file
MAC-Oriented CFG file
MAC-local CFG file
Common CFG file
A Common CFG file contains parameters that affect the basic operation of the IP phone, such as language and volume. It will be effectual for all IP phones of the same model.
The common CFG file has a fixed name for each IP phone model. The name of the
Common CFG file for each SIP-T4X IP phone model is:
SIP-T48G: y000000000035.cfg
SIP-T46G: y000000000028.cfg
SIP-T42G: y000000000029.cfg
SIP-T41P: y000000000036.cfg
MAC-Oriented CFG file
A MAC-Oriented CFG file contains parameters unique to a particular phone. It will only be effectual for a specific IP phone. The MAC-Oriented CFG file is named after the MAC address of the IP phone. For example, if the MAC address of an IP phone is
001565113af8, the name of the MAC-Oriented CFG file must be 001565113af8.cfg.
MAC-local CFG file
A MAC-local CFG file contains changes that users make via web user interface and phone user interface. It will only be effectual for a specific IP phone. The MAC-local CFG
17
Administrator’s Guide for SIP-T4X IP Phones file is named after the MAC address of the IP phone. This file is stored locally on the IP phone and can also be uploaded to the provisioning server.
The MAC-local CFG file enables the phone to protect personalized settings. For more information on how to protect personalized settings, refer to
the section
Specific
Scenarios-Protect Personalized Settings
in
Yealink_SIP-T2_Series_T4_Series_IP_Phones_Auto_Provisioning_Guide
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
Central Provisioning
IP phones can be centrally provisioned from a provisioning server using the configuration files (<y0000000000xx>.cfg and <MAC>.cfg). You can use a text-based editing application to edit configuration files, and then store configuration files to a
provisioning server. For more information on the provisioning server, refer to Provisioning
IP phones can obtain the provisioning server address during startup. 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_T4_Series_IP_Phones_Auto_Provisioning_Guide
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
When modifying parameters, learn the following:
Parameters in configuration files override those stored in IP phones’ flash memory.
The .cfg extension of the configuration files must be in lowercase.
Each line in a configuration file must use the following format and adhere to the following rules: variable-name = value
-
Associate only one value with one variable.
- Separate variable name and value with equal sign.
-
Set only one variable per line.
-
Put the variable and value on the same line, and do not break the line.
-
Comment the variable on a separated line. Use the pound (#) delimiter to distinguish the comments.
18
IP phones perform the auto provisioning function of downloading configuration files,
Getting Started downloading resource files and upgrading firmware. The transfer protocol is used to download files from the provisioning server. IP phones support several transport protocols for provisioning, including FTP, TFTP, HTTP, and HTTPS protocols. And you can specify the transport protocol in the provisioning server address, for example, http://xxxxxxx. If not specified, the TFTP protocol is used. The provisioning server address can be IP address, domain name or URL. If a user name and password are specified as part of the provisioning server address, for example, http://user:pwd@/server/dir, they will be used only if the server supports them.
Note
A URL should contain forward slashes instead of back slashes and should not contain spaces. Escape characters are not supported.
If a user name and password are not specified as part of the provisioning server address, the User Name and Password of the provisioning server configured on the IP phone will be used.
There are two types of FTP methods—active and passive. IP phones are not compatible with active FTP.
The provisioning server can be on the local LAN or anywhere on the Internet. Use the following procedure as a recommendation if this is your first provisioning server setup.
For more information on how to set up a provisioning server, refer to
Yealink_SIP-T2_Series_T4_Series_IP_Phones_Auto_Provisioning_Guide
.
To set up the provisioning server:
1. Install a provisioning server application or locate a suitable existing server.
2. Create an account and home directory.
3. Set security permissions for the account.
4. Create configuration files and edit them as desired.
5. Copy the configuration files and resource files to the provisioning server.
For more information on how to deploy IP phones using configuration files, refer to
Deploying Phones from the Provisioning Server on page 19 .
Note
Typically all phones are configured with the same server account, but the server account provides a means of conveniently partitioning the configuration. Give each account a unique home directory on the server and change the configuration on a per-account basis.
The parameters in the new downloaded configuration files will override the duplicate parameters in files downloaded earlier. During auto provisioning, IP phones download
19
Administrator’s Guide for SIP-T4X IP Phones the common configuration file first, and then the MAC-oriented file. Therefore any parameter in the MAC-oriented configuration file will override the same one in the common configuration file.
Yealink supplies configuration files for each phone model, which is delivered with the IP phone firmware. The configuration files, supplied with each firmware release, must be used with that release. Otherwise, configurations may not take effect, and the IP phone will behave without exception. Before you configure parameters in the configuration files, Yealink recommends that you create new configuration files containing only those parameters that require changes.
To deploy IP phones from the provisioning server:
1. Create per-phone configuration files by performing the following steps: a) Obtain a list of phone MAC addresses (the bar code label on the back of the
IP phone or on the outside of the box). b) Create per-phone <MAC>.cfg files by using the MAC-Oriented CFG file from the distribution as templates. c) Edit the parameters in the file as desired.
2. Create new common configuration files by performing the following steps: a) Create <y0000000000xx>.cfg files by using the Common CFG file from the distribution as templates. b) Edit the parameters in the file as desired.
3. Copy configuration files to the home directory of the provisioning server.
4. Reboot IP phones to trigger the auto provisioning process.
IP phones discover the provisioning server address, and then download the configuration files from the provisioning server.
For more information on configuration files, refer to Configuration Files on page 17 . For
more information on encrypting configuration files, refer to Encrypting Configuration
During the auto provisioning process, the IP phone supports the following methods to discover the provisioning server address:
Zero Touch: Zero Touch feature guides you to configure network settings and the provisioning server address via phone user interface after startup.
PnP: PnP feature allows IP phones to discover the provisioning server address by broadcasting the PnP SUBSCRIBE message during startup.
DHCP: DHCP option can be used to provide the address or URL of the provisioning server to IP phones. When the IP phone requests an IP address using the DHCP protocol, the resulting response may contain option 66 or the custom option (if configured) that contains the provisioning server address.
Static: You can manually configure the server address via phone user interface or web user interface.
20
Getting Started
For more information on the above methods, refer to
Yealink_SIP-T2_Series_T4_Series_IP_Phones_Auto_Provisioning_Guide
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
In order to get your IP phones running, you must perform basic network setup, such as IP address and subnet mask configuration. This section describes how to configure basic network parameters for IP phones.
Note
This section mainly introduces IPv4 network parameters. IP phones also support IPv6. For more information on IPv6, refer to
on page
DHCP (Dynamic Host Configuration Protocol) is a network protocol used to dynamically allocate network parameters to network hosts. The automatic allocation of network parameters to hosts eases the administrative burden of maintaining an IP network. IP phones comply with the DHCP specifications documented in RFC 2131. If DHCP is used,
IP phones connected to the network become operational without having to be manually assigned IP addresses and additional network parameters. Static DNS address(es) can be configured and used when DHCP is enabled.
DHCP Option
DHCP provides a framework for passing information to TCP/IP network devices. Network and other control information are carried in tagged data items that are stored in the options field of the DHCP message. The data items themselves are also called options.
DHCP can be initiated by simply connecting the IP phone with the network. IP phones broadcast DISCOVER messages to request the network information carried in DHCP options, and the DHCP server responds with the specific values in the corresponding options.
The following table lists the common DHCP options supported by IP phones.
Parameter
Subnet Mask
Time Offset
Router
DHCP Option
1
2
3
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.
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Administrator’s Guide for SIP-T4X IP Phones
Parameter
Time Server
Domain Name
Server
Log Server
Host Name
Domain Server
DHCP Option
4
6
7
12
15
Description
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
28
42
43
60
Specify a list of the NTP servers available to the client by IP address.
Identify the vendor-specific information.
Identify the vendor type.
TFTP Server
Name
Bootfile Name
66
67
Identify a TFTP server when the 'sname' field in the DHCP header has been used for DHCP options.
Identify a bootfile when the 'file' field in the
DHCP header has been used for DHCP options.
For more information on DHCP options, refer to http://www.ietf.org/rfc/rfc2131.txt?number=2131 or http://www.ietf.org/rfc/rfc2132.txt?number=2132 .
If you do not have the ability to configure the DHCP options for discovering the provisioning server on the DHCP server, an alternate method of automatically discovering the provisioning server address is required. Connecting to the secondary
DHCP server that responds to DHCP INFORM queries with a requested provisioning server address is one possibility. For more information, refer to http://www.ietf.org/rfc/rfc3925.txt?number=3925 .
22
Getting Started
Procedure
DHCP can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure DHCP on the IP phone.
Parameter: network.internet_port.type
Configure static DNS address when DHCP is used.
Parameters: network.primary_dns network.secondary_dns
Configure the IP phone to use manually configured static IPv4
DNS.
Parameters: network.static_dns_enable
Configure DHCP on the IP phone.
Configure static DNS address when DHCP is used.
Navigate to: http://<phoneIPAddress>/servlet
?p=network&q=load
Configure DHCP on the IP phone.
Configure static DNS address when DHCP is used.
Details of Configuration Parameters:
Parameters Permitted Values Default network.internet_port.type 0, 1 or 2 0
Description:
Configures the Internet (WAN) port type for IPv4 when the IP address mode is configured as IPv4 or IPv4&IPv6.
0-DHCP
1-PPPoE (not applicable to SIP-T42G/T41P)
2-Static IP Address
Note: If you change this parameter, the IP phone will reboot to make the change take
23
Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values Default effect.
Web User Interface:
Network->Basic->IPv4 Config
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4->Type network.static_dns_enable 0 or 1 0
Description:
Enables or disables the IP phone to use manually configured static IPv4 DNS when the Internet (WAN) port type for IPv4 is configured as DHCP.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->Static DNS
Phone User Interface:
Menu->Advanced (default password: admin)->Network->WAN Port->IPv4->Static
DNS network.primary_dns IPv4 Address Blank
Description:
Configures the primary IPv4 DNS server when the static IPv4 DNS is enabled.
Example: network.primary_dns = 202.101.103.55
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->Static IP Address->Primary DNS
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4->Static
DNS (Enabled)->Primary DNS network.secondary_dns IPv4 Address Blank
24
Getting Started
Parameters Permitted Values Default
Description:
Configures the secondary IPv4 DNS server when the static IPv4 DNS is enabled.
Example: network.secondary_dns = 202.101.103.54
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->Static IP Address->Secondary DNS
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4->Static
DNS (Enabled)->Secondary DNS
To configure DHCP via web user interface:
1. Click on Network->Basic.
2. In the IPv4 Config block, mark the DHCP radio box.
3. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after reboot.
4. Click OK to reboot the 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.
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Administrator’s Guide for SIP-T4X IP Phones
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 phone.
To configure DHCP via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4.
2. Press or , or the Switch soft key to select the DHCP from the Type field.
3. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
To configure static DNS when DHCP is used via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv4->Static DNS.
2. Press or , or the Switch soft key to select Enabled from the Static DNS field.
3. Enter the desired values in the Primary DNS and Secondary DNS fields respectively.
4. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
26
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
Getting Started
Subnet Mask
Default Gateway
Primary DNS
Secondary DNS
Procedure
Network parameters can be configured manually using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Phone User Interface
Configure network parameters of the IP phone manually.
Parameters: network.internet_port.type network.ip_address_mode network.internet_port.ip network.internet_port.mask network.internet_port.gateway network.primary_dns network.secondary_dns
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.
Details of Configuration Parameters:
Parameters Permitted Values Default network.internet_port.type 0, 1 or 2 0
Description:
Configures the Internet (WAN) port type for IPv4 when the IP address mode is configured as IPv4 or IPv4&IPv6.
0-DHCP
1-PPPoE (not applicable to SIP-T42G/T41P)
2-Static IP Address
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values Default
Web User Interface:
Network->Basic->IPv4 Config
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4->Type network.ip_address_mode 0, 1 or 2 0
Description:
Configures the IP address mode.
0-IPv4
1-IPv6
2-IPv4&IPv6
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->Internet Port->Mode (IPv4/IPv6)
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN Port->IP Address
Mode network.internet_port.ip IPv4 Address Blank
Description:
Configures the IPv4 address when the IP address mode is configured as IPv4 or
IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP
Address.
Example: network.internet_port.ip = 192.168.1.20
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->Static IP Address->IP Address
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv4->Type(Static IP)->IP Address network.internet_port.mask Subnet Mask Blank
28
Getting Started
Parameters Permitted Values Default
Description:
Configures the IPv4 subnet mask when the IP address mode is configured as IPv4 or
IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP
Address.
Example: network.internet_port.mask = 255.255.255.0
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->Static IP Address->Subnet Mask
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv4->Type(Static IP)->Subnet Mask network.internet_port.gateway IPv4 Address Blank
Description:
Configures the IPv4 default gateway when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP
Address.
Example: network.internet_port.gateway = 192.168.1.254
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->Static IP Address->Gateway
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv4->Type(Static IP)->Gateway network.primary_dns IPv4 Address Blank
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Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values Default
Description:
Configures the primary IPv4 DNS server when the IP address mode is configured as
IPv4 or IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP
Address.
Example: network.primary_dns = 202.101.103.55
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->Static IP Address->Primary DNS
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv4->Type(Static IP)->Primary DNS network.secondary_dns IPv4 Address Blank
Description:
Configures the secondary IPv4 DNS server when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static
IP Address.
Example: network.secondary_dns = 202.101.103.54
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->Static IP Address->Secondary DNS
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv4->Type(Static IP)->Secondary DNS
To configure the IP address mode via web user interface:
1. Click on Network->Basic.
30
2. Select the desired value from the pull-down list of Mode (IPv4/IPv6).
Getting Started
3. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after reboot.
4. Click OK to reboot the phone.
To configure a static IPv4 address via web user interface:
1. Click on Network->Basic.
2. In the IPv4 Config block, mark the Static IP Address radio box.
3. Enter the IP address, subnet mask, default gateway, primary DNS and secondary
DNS in the corresponding fields.
4. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after reboot.
5. Click OK to reboot the phone.
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Administrator’s Guide for SIP-T4X IP Phones
To configure the IP address mode via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->WAN Port.
2. Press or to highlight the IP Address Mode field.
3. Press or to select IPv4 or IPv4&IPv6 from the IP Address Mode field.
4. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
To configure a static IPv4 address via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4.
2. Press or , or the Switch soft key to select the Static IP from the Type field.
3. Enter the desired values in the IP Address, Subnet Mask, Gateway, Primary DNS and Secondary DNS fields respectively.
4. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
32
PPPoE (Point-to-Point Protocol over Ethernet) is a network protocol used by Internet
Service Providers (ISPs) to provide Digital Subscriber Line (DSL) high speed Internet services. PPPoE allows an office or building-full of users to share a common DSL connection to the Internet. PPPoE connection is supported by the Internet port of the IP phone. Contact your ISP for the PPPoE user name and password. PPPoE is not applicable to SIP-T42G and SIP-T41P IP phones.
Procedure
PPPoE can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Configure PPPoE on the IP phone.
Parameters: network.internet_port.type
Configure the user name and password for PPPoE on the IP phone.
Parameters: network.pppoe.user network.pppoe.password
Configure PPPoE on the IP phone.
Navigate to:
Getting Started
Phone User Interface http://<phoneIPAddress>/servlet
?p=network&q=load
Configure PPPoE on the IP phone.
Details of Configuration Parameters:
Parameters Permitted Values Default network.internet_port.type 0, 1 or 2 0
Description:
Configures the Internet (WAN) port type for IPv4 when the IP address mode is configured as IPv4 or IPv4&IPv6.
0-DHCP
1-PPPoE (not applicable to SIP-T42G/T41P)
2-Static IP Address
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4->Type network.pppoe.user String within 32 characters Blank
Description:
Configures the user name for PPPoE connection when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet port type is configured as PPPoE.
Example: network.pppoe.user = Xmyl05921234
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->PPPoE->User
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv4->Type(PPPoE) ->PPPoE User network.pppoe.password String within 99 characters Blank
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Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values Default
Description:
Configures the password for PPPoE connection when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet port type is configured as PPPoE.
Example: network.pppoe.password = yealink123
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv4 Config->PPPoE->Password
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv4->Type(PPPoE) ->PPPoE Password
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 the corresponding fields.
34
4. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after reboot.
5. Click OK to reboot the phone.
To configure PPPoE via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4.
2. Press or , or the Switch soft key to select the PPPoE from the Type field.
3. Enter the user name and password in the corresponding fields.
Getting Started
4. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
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 all connected devices choose common transmission parameters (e.g., speed and duplex mode) to transmit voice or data over Ethernet. This process entails devices first sharing transmission capabilities and then selecting the highest performance transmission mode supported by both. You can configure the
Internet port and PC port on IP phones to auto-negotiate during the transmission.
Half-duplex
Half-duplex transmission refers to transmitting voice or data in both directions, but in one direction at a time; this means one device can send data on the line, but not receive data simultaneously. You can configure the half-duplex transmission on both
Internet port and PC port for IP phones to transmit in 10Mbps or 100Mbps.
35
Administrator’s Guide for SIP-T4X IP Phones
Full-duplex
Full-duplex transmission refers to transmitting voice or data in both directions at the same time; this means one device can send data on the line while receiving data. You can configure the full-duplex transmission on both Internet port and PC port for IP phones to transmit in 10Mbps, 100Mbps or 1000Mbps (1000Mbps is not applicable to
SIP-T41P).
36
Procedure
The transmission method of Ethernet port can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Local Web User Interface
Configure the transmission methods of Ethernet ports.
Parameters: network.internet_port.speed_duplex network.pc_port.speed_duplex
Configure the transmission method of Ethernet port.
Navigate to: http://<phoneIPAddress>/servlet?p
=network-adv&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default network.internet_port.speed_duplex 0, 1, 2, 3, 4 or 5 0
Getting Started
Parameters Permitted Values Default
Description:
Configures the transmission method and speed of the Internet (WAN) port.
0-Auto negotiate
1-Full duplex, 10Mbps
2-Full duplex, 100Mbps
3-Half duplex, 10Mbps
4-Half duplex, 100Mbps
5-Full duplex, 1000Mbps (not applicable to SIP-T41P)
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.
Web User Interface:
Network->Advanced->Port Link->WAN Port Link
Phone User Interface:
None network.pc_port.speed_duplex 0, 1, 2, 3, 4 or 5 0
Description:
Configures the transmission method and speed of the PC (LAN) port.
0-Auto negotiate
1-Full duplex, 10Mbps
2-Full duplex, 100Mbps
3-Half duplex, 10Mbps
4-Half duplex, 100Mbps
5-Full duplex, 1000Mbps (not applicable to SIP-T41P)
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.
Web User Interface:
Network->Advanced->Port Link->PC Port Link
Phone User Interface:
None
To configure the transmission method of Ethernet port via web user interface:
1. Click on Network->Advanced.
2. Select the desired value from the pull-down list of WAN Port Link.
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Administrator’s Guide for SIP-T4X IP Phones
3. Select the desired value from the pull-down list of PC Port Link.
38
4. Click Confirm to accept the change.
The PC port on the back of the IP phone is used to connect a PC. You can enable or disable the PC (LAN) port on SIP-T4X IP phones via web user interface or using configuration files.
Procedure
PC port can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the PC port.
Parameters: network.PC_port.enable
Configure the PC port.
Navigate to: http://<phoneIPAddress>/servlet
?p=network-pcport&q=load
Details of Configuration Parameters:
Parameter Permitted Values Default network.PC_port.enable 0 or 1 1
Getting Started
Parameter Permitted Values Default
Description:
Enables or disables the PC (LAN) port.
0-Disabled
1-Auto Negotiation
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->PC Port->PC Port Active
Phone User Interface:
None
To enable the PC port via web user interface:
1. Click on Network->PC Port.
2. Select Auto Negotiation from the pull-down list of PC Port Active.
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 phone.
To disable the PC port via web user interface:
1. Click on Network->PC Port.
2. Select Disabled from the pull-down list of PC Port Active.
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Administrator’s Guide for SIP-T4X 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 phone.
This section provides information on 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-T48G
SIP-T46G
SIP-T42G
SIP-T41P
Associated Firmware Name
35.x.x.x.rom
28.x.x.x.rom
29.x.x.x.rom
36.x.x.x.rom
Firmware Name Example
35.73.0.40.rom
28.73.0.40.rom
29.73.0.40.rom
36.73.0.40.rom
Note
You can download the latest firmware online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
Do not unplug the network and power cables when the IP phone is upgrading firmware.
Upgrade via Web User Interface
To manually upgrade firmware via web user interface, you need to store the firmware to the local system in advance.
To upgrade firmware manually via web user interface:
1. Click on Settings->Upgrade.
2. Click Browse.
3. Locate the firmware from the local system.
4. Click Upgrade.
40
Getting Started
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 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 using the 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 booting up.
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.
Parameters: auto_provision.power_on auto_provision.repeat.enable auto_provision.repeat.minutes auto_provision.weekly.enable auto_provision.weekly.begin_time
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Administrator’s Guide for SIP-T4X IP Phones
Local Web User Interface auto_provision.weekly.end_time auto_provision.weekly.dayofweek
Specify the access URL of firmware.
Parameter: firmware.url
Configure the way for the IP phone to check for configuration files.
Navigate to: http://<phoneIPAddress>/servlet?p
=settings-autop&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default auto_provision.power_on 0 or 1 1
Description:
Enables or disables the IP phone to perform an auto provisioning process when powered on.
0-Disabled
1-Enabled
Web User Interface:
Settings->Auto Provision->Power On
Phone User Interface:
None auto_provision.repeat.enable 0 or 1 0
Description:
Enables or disables the IP phone to perform an auto provisioning process repeatedly.
0-Disabled
1-Enabled
Web User Interface:
Settings->Auto provision->Repeatedly
Phone User Interface:
None
42
Getting Started
Parameters Permitted Values Default auto_provision.repeat.minutes Integer from 1 to 43200 1440
Description:
Configures the interval (in minutes) for the IP phone to perform an auto provisioning process repeatedly.
Note: It works only if the value of the parameter “auto_provision.repeat.enable” is set to 1(Enabled).
Web User Interface:
Settings->Auto provision->Interval (Minutes)
Phone User Interface:
None auto_provision.weekly.enable 0 or 1 0
Description:
Enables or disables the IP phone to perform an auto provisioning process weekly.
0-Disabled
1-Enabled
Web User Interface:
Settings->Auto provision->Weekly
Phone User Interface:
None auto_provision.weekly.begin_time Time from 00:00 to 23:59 00:00
Description:
Configures the begin time of the day for the IP phone to perform an auto provisioning process weekly.
Note: It works only if the value of the parameter “auto_provision.weekly.enable” is set to 1(Enabled).
Web User Interface:
Settings->Auto provision->Time
Phone User Interface:
None auto_provision.weekly.end_time Time from 00:00 to 23:59 00:00
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Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values Default
Description:
Configures the end time of the day for the IP phone to perform an auto provisioning process weekly
.
Note: It works only if the value of the parameter “auto_provision.weekly.enable” is set to 1(Enabled).
Web User Interface:
Settings->Auto provision->Time
Phone User Interface:
None auto_provision.weekly.dayofweek
0,1,2,3,4,5,6 or a combination of these digits
0123456
Description:
Configures the days of the week for the IP phone to perform an auto provisioning process weekly.
0-Sunday
1-Monday
2-Tuesday
3-Wednesday
4-Thursday
5-Friday
6-Saturday
Example: auto_provision.weekly.dayofweek = 01 means the IP phone will perform an auto provisioning process every Sunday and Monday.
Note: It works only if the value of the parameter “auto_provision.weekly.enable” is set to 1(Enabled). The old parameters “auto_provision.weekly.mask” is also applicable to SIP-T4X IP phones.
Web User Interface:
Settings->Auto provision->Day of Week
Phone User Interface:
None firmware.url URL within 511 characters Blank
44
Getting Started
Parameters Permitted Values Default
Description:
Configures the access URL of the firmware file.
Example: firmware.url = http://192.168.1.20/2.73.0.40.rom
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Settings->Upgrade->Select and Upgrade Firmware
Phone User Interface:
None
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.
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Administrator’s Guide for SIP-T4X IP Phones
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.
46
Configuring Basic Features
This chapter provides information for making configuration changes for the following basic features:
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Administrator’s Guide for SIP-T4X IP Phones
Transfer on Conference Hang Up
Calling Line Identification Presentation
Connected Line Identification Presentation
48
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.
Configuring Basic Features
Ringing Power Light Flash
Ringing Power Light Flash allows the power indicator LED to flash when the IP phone receives an incoming call.
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.
Mute Power Light Flash
Mute Power Light Flash allows the power indicator LED to flash when a call is mute.
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.
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.
Note
Power indicator LED feature is only applicable to IP phones running firmware version 72 or later.
Procedure
Power indicator LED can be configured using the configuration files or locally.
Configuration File
<y0000000000xx>.cf
g
Local Web User Interface
Configure the power indicator LED.
Parameters: phone_setting.common_power_led_e
nable phone_setting.ring_power_led_flash_
enable phone_setting.mail_power_led_flash_
enable phone_setting.mute_power_led_flash_
enable phone_setting.hold_and_held_power_
led_flash_enable phone_setting.talk_and_dial_power_l
ed_enable
Configure the power indicator LED.
Navigate to:
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Administrator’s Guide for SIP-T4X IP Phones http://<phoneIPAddress>/servlet?p=f eatures-powerled&q=load
Details of Configuration Parameters:
Parameters
Permitted
Values
0 or 1
Default
0 phone_setting.common_power_led_enable
Description:
Enables or disables the power indicator LED to be turned on.
0-Disabled (power indicator LED is off)
1-Enabled (power indicator LED is solid red)
Note: The old parameter “features.power_led_on” is also applicable to IP phones and “features.idle_talk_power_led_flash_enable” is also applicable to SIP-T4X IP phones.
Web User Interface:
Features->Power LED->Common Power Light On
Phone User Interface:
None phone_setting.ring_power_led_flash_enable 0 or 1 1
Description:
Enables or disables the power indicator LED to flash when the IP phone receives an incoming call.
0-Disabled (power indicator LED does not flash)
1-Enabled (power indicator LED fast flashes (300ms) red)
Web User Interface:
Features->Power LED->Ringing Power Light Flash
Phone User Interface:
None phone_setting.mail_power_led_flash_enable 0 or 1 1
Description:
Enables or disables the power indicator LED to flash when the IP phone receives a voice mail or a text message.
0-Disabled (power indicator LED does not flash)
1-Enabled (power indicator LED slow flashes (1000ms) red)
50
Configuring Basic Features
Parameters
Web User Interface:
Features->Power LED->Voice/Text Mail Power Light Flash
Phone User Interface:
None phone_setting.mute_power_led_flash_enable
Permitted
Values
0 or 1
Description:
Enables or disables the power indicator LED to flash when a call is mute.
0-Disabled (power indicator LED does not flash)
1-Enabled (power indicator LED fast flashes (300ms) red)
Web User Interface:
Features->Power LED->Mute Power Light Flash
Phone User Interface:
None phone_setting.hold_and_held_power_led_flash_enable 0 or 1
Default
0
0
Description:
Enables or disables the power indicator LED to flash when a call is placed on hold or is held.
0-Disabled (power indicator LED does not flash)
1-Enabled ( power indicator LED fast flashes (500ms) red)
Web User Interface:
Features->Power LED->Hold/Held Power Light Flash
Phone User Interface:
None phone_setting.talk_and_dial_power_led_enable 0 or 1 0
Description:
Enables or disables the power indicator LED to be turned on when the IP phone is busy.
0-Disabled (power indicator LED is off)
1-Enabled (power indicator LED is solid red)
Web User Interface:
Features->Power LED->Talk/Dial Power Light On
Phone User Interface:
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Administrator’s Guide for SIP-T4X IP Phones
Parameters
Permitted
Values
Default
None
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 Ringing 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.
7. Select the desired value from the pull-down list of Talk/Dial Power Light On.
8. Click Confirm to accept the change.
Notification popups feature allows the IP phone to display the pop-up message when it misses a call, forwards an incoming call to other party or receives a new voice mail or a new text message.
Note
Notification popups feature is applicable to IP phones running firmware version 73 or later.
52
Configuring Basic Features
Procedure
Notification popups can be configured using the configuration files or locally.
Configuration File
<y0000000000xx>.cf
g
Local Web User Interface
Configure notification popups.
Parameters: features.voice_mail_popup.enable features.missed_call_popup.enable features.forward_call_popup.enable features.text_message_popup.enable
Configure notification popups.
Navigate to: http://<phoneIPAddress>/servlet?p=f eatures-notifypop&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default features.voice_mail_popup.enable 0 or 1 1
Description:
Enables or disables the IP phone to display the pop-up message box when it receives a new voice mail.
0-Disabled
1-Enabled
Note: If the voice mail pop-up message box disappears, it won't pop up again unless the user receives a new voice mail or the user re-registers the account that has unread voice mail(s). It is only applicable to IP phones running firmware version 73 or later.
Web User Interface:
Features->Notification Popups->Display Voice Mail Popup
Phone User Interface:
None features.missed_call_popup.enable 0 or 1 1
Description:
Enables or disables the IP phone to display the pop-up message box when it misses a call.
0-Disabled
1-Enabled
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Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values
Note: It is only applicable to IP phones running firmware version 73 or later.
Web User Interface:
Features->Notification Popups->Display Missed Call Popup
Phone User Interface:
None
Default features.forward_call_popup.enable 0 or 1 1
Description:
Enables or disables the IP phone to display the pop-up message box when it forwards an incoming call to other party.
0-Disabled
1-Enabled
Note: It is only applicable to IP phones running firmware version 73 or later.
Web User Interface:
Features->Notification Popups->Display Forward Call Popup
Phone User Interface:
None features.text_message_popup.enable 0 or 1 1
Description:
Enables or disables the IP phone to display the pop-up message box when it receives a new text message.
0-Disabled
1-Enabled
Note: It is only applicable to SIP-T46G IP phones running firmware version 73 or later.
Web User Interface:
Features->Notification Popups->Display Text Message Popup
Phone User Interface:
None
To configure the notification popups via web user interface:
1. Click on Features->Notification Popups.
2. Select the desired value from the pull-down list of Display Voice Mail Popup.
3. Select the desired value from the pull-down list of Display Missed Call Popup.
4. Select the desired value from the pull-down list of Display Forward Call Popup.
54
Configuring Basic Features
5. Select the desired value from the pull-down list of Display Text Message Popup.
6. Click Confirm to accept the change.
Contrast determines the readability of the texts displayed on the LCD screen. Adjusting the contrast to a comfortable level can optimize the screen viewing experience. When configured properly, contrast allows users to read the LCD’s display with minimal eyestrain. You can only configure the LCD’s contrast of the expansion module EXP40 connected to SIP-T48G and T46G IP phones. Make sure the expansion module has been connected to the IP phone before adjustment.
Procedure
Contrast can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Phone User Interface
Configures the LCD’s contrast of
EXP40 connected to
SIP-T48G/T46G IP phones.
Parameter: phone_setting.contrast
Configures the LCD’s contrast of
EXP40 connected to
SIP-T48G/T46G IP phones.
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Administrator’s Guide for SIP-T4X IP Phones
Details of the Configuration Parameter:
Parameter Permitted Values Default phone_setting.contrast Integer from 1 to 10 6
Description:
Configures the LCD’s contrast of EXP40 connected to SIP-T48G/T46G IP phones.
Note: We recommend that you set the contrast of the LCD screen to 6 as a more comfortable level.
Web User Interface:
None
Phone User Interface:
Menu->Basic->Display->Contrast
To configure contrast via phone user interface (only applicable to EXP40 connected to
SIP-T48G/T46G IP phones):
1. Press Menu->Basic->Display->Contrast.
2. Press or , or the Switch soft key to increase or decrease the intensity of contrast.
The default contrast level is 6.
3. Press the Save soft key to accept the change.
Wallpaper is an image used as the background of the IP phone idle screen. Users can select an image from IP phone’s built-in background or customize wallpaper from personal pictures. To set the custom wallpaper as the IP phone background, you need to upload the custom wallpaper to the IP phone in advance. The wallpaper is not applicable to SIP-T42G and SIP-T41P IP phones.
The wallpaper image format must meet the following:
Phone Model
SIP-T46G
SIP-T48G
Format Resolution Single File Size Total File Size
.jpg/.png/.bmp <=480*272 <=5MB <=20MB
.jpg/.png/.bmp <=800*480 <=5MB <=20MB
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Configuring Basic Features
Procedure
Wallpaper can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configures the wallpaper displayed on the IP phone.
Parameter: phone_setting.backgrounds
Specify the access URL of the custom wallpaper.
Parameter: wallpaper_upload.url
Upload the custom wallpaper.
Change the wallpaper via web user interface.
Navigate to: http://<phoneIPAddress>/se rvlet?p=settings-preference
&q=load
Change the wallpaper via phone user interface.
Details of the Configuration Parameter:
Parameters Permitted Values Default phone_setting.backgrounds
Refer to the following content
Resource: Default.png
Description:
Configures the wallpaper displayed on the IP phone.
Example:
To set a phone built-in picture (e.g., 1.png) to be wallpaper, the value format is: phone_setting.backgrounds = Resource:1.png
To configure a custom picture (e.g., custom1.png) to be wallpaper, the value format is:
Config:custom1.png
Permitted Values:
Resource:X (Valid values of X are: Default.png,1.png,2.png,3.png,4.png,5.png, 6.png,
7.png,8.png or 9.png) or Config:wallpaper name
Note: It is only applicable to SIP-T48G/T46G IP phones.
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Administrator’s Guide for SIP-T4X IP Phones
Parameters
Web User Interface:
Settings->Preference->Wallpaper
Phone User Interface:
Menu->Basic->Display->
Wallpaper
Permitted Values wallpaper_upload.url
URL within 511 characters
Default
Blank
Description:
Configures the access URL of the wallpaper image.
Example: wallpaper_upload.url = http://192.168.10.25/wallpaper.jpg
Note: It is only applicable to SIP-T48G/T46G IP phones.
Web User Interface:
Settings->Preference->Upload Wallpaper (480*272)
Phone User Interface:
None
To upload custom wallpaper via web user interface:
1. Click on Settings->Preference.
2. In the Upload Wallpaper field, click Browse to locate the wallpaper image from your local system.
3. Click Upload to upload the file.
58
4. Click Confirm to accept the change.
The custom wallpaper appears in the pull-down list of Wallpaper.
Configuring Basic Features
To change the wallpaper via web user interface:
1. Click on Settings->Preference.
2. Select the desired wallpaper from the pull-down list of Wallpaper.
3. Click Confirm to accept the change.
To change the wallpaper via phone user interface:
1. Press Menu->Basic->Display->Wallpaper.
2. Press or , or the Switch soft key to select the desired wallpaper.
3. Press the Save soft key to accept the change.
Backlight determines the brightness of the LCD screen display, allowing users to read easily in dark environments. Backlight time specifies the delay time to change the backlight when the IP phone is inactive. Backlight turns off quickly if a short backlight time is configured, this may not give users enough time to read messages. Backlight time is applicable to SIP-T4X IP phones and EXP40 connected to SIP-T48G/T46G IP phones.
You can configure the backlight time as one of the following types:
Always On: Backlight is turned on permanently.
15s, 30s, 60s, 120s, 300s, 600s or 1800s: Backlight is turned off or turned dusky when the IP phone is inactive after a preset period of time. It is automatically turned on if the status of the IP phone changes or any key is pressed.
Backlight Active Level is used to adjust the backlight intensity of the LCD screen when the phone is active. Backlight Inactive Level is used to adjust the backlight intensity of the LCD screen when the phone is inactive. Backlight Active Level is only applicable to
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Administrator’s Guide for SIP-T4X IP Phones
SIP-T48G and SIP-T46G IP phones and the connected EXP40. Backlight Inactive Level is only applicable to SIP-T48G and SIP-T46G IP phones.
Procedure
Backlight can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cf
g
Web User Interface
Phone User Interface
Configure the backlight of the LCD screen.
Parameters: phone_setting.active_backlight_level phone_setting.inactive_backlight_level phone_setting.backlight_time
Configure the backlight of the LCD screen.
Navigate to: http://<phoneIPAddress>/servlet?p=s ettings-preference&q=load
Configure the backlight of the LCD screen.
Details of Configuration Parameters:
Parameters Permitted Values Default phone_setting.active_backlight_level Integer from 1 to 10 8
Description:
Configures the intensity of the LCD screen when the phone is active.
10 is the highest intensity.
Note: It is only applicable to SIP-T48G/T46G IP phones and the connected EXP40.
Web User Interface:
Settings->Preference->Backlight Active Level
Phone User Interface:
Menu->Basic->Display->Backlight->Backlight Active Level phone_setting.inactive_backlight_level 0 or 1 1
Description:
Configures the intensity of the LCD screen when the phone is inactive.
0-Off
60
Configuring Basic Features
Parameters Permitted Values
1-Low
Note: It is only applicable to SIP-T48G/T46G IP phones.
Web User Interface:
Settings->Preference->Backlight Inactive Level
Phone User Interface:
Menu->Basic->Display->Backlight->Backlight Inactive Level phone_setting.backlight_time
0, 15, 30, 60, 120, 300, 600 or
1800
Default
0
Description:
Configures the delay time (in seconds) to change the intensity of the LCD screen when the IP phone is inactive.
0-Always on
15-15s
30-30s
60-60s
120-120s
300-300s
600-600s
1800-1800s
If it is set to 60 (60s), the intensity of the LCD screen will be changed when the IP phone is inactive for 60 seconds.
Web User Interface:
Settings->Preference->Backlight Time(seconds)
Phone User Interface:
Menu->Basic->Display->Backlight->Backlight Time
To configure the backlight via web user interface:
1. Click on Settings->Preference.
2. Select the desired value from the pull-down list of Backlight Inactive Level.
3. Select the desired value from the pull-down list of Backlight Active Level.
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Administrator’s Guide for SIP-T4X IP Phones
4. Select the desired value from the pull-down list of Backlight Time (seconds).
62
5. Click Confirm to accept the change.
To configure the backlight via phone user interface:
1. Press Menu->Basic->Display->Backlight.
2. Press or , or the Switch soft key to select the desired level from the
Backlight Active Level field.
3. Press or , or the Switch soft key to select the desired value from the
Backlight Inactive Level field.
4. Press or , or the Switch soft key to select the desired time from the
Backlight Time field.
5. Press the Save soft key to accept the change.
Display contact photo allows the IP phone to present the contact avatar when it receives an incoming call, dials an outgoing call or engages in a call. Display contact photo feature is only applicable to SIP-T48G/SIP-T46G IP phones.
Display Called Party Information allows the IP phone to present the callee identity in addition to the presentation of caller identity when it receives an incoming call.
Configuring Basic Features
The following figure shows an incoming call from Yealink 4604 to Yealink 4603 on the
SIP-T46G IP phone.
You can customize the call information to be displayed on the IP phone as required. IP phones support five call information display methods: Number+Name, Name,
Name+Number, Number and Full Contact Info (display name<sip:[email protected]>).
Note
SIP-T42G/T41P IP phones have a limited display (up to three lines) due to their smaller screen size.
Procedure
Web server type can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure display contact photo feature.
Parameter: phone_setting.contact_photo_d
isplay.enable
Configure display called party information feature.
Parameter: phone_setting.called_party_inf
o_display.enable
Configure the caller information display method.
Parameter: phone_setting.call_info_display
_method
Configure call display feature.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-calldisplay&q=lo
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Administrator’s Guide for SIP-T4X IP Phones ad
Details of Configuration Parameters:
Parameters Permitted Values Default phone_setting.contact_photo_display.enable 0 or 1 1
Description:
Enables or disables the IP phone to display contact avatar when it receives an incoming call, dials an outgoing call or engages in a call.
0-Disabled
1-Enabled
Note: It is only applicable to SIP-T48G/T46G IP phones.
Web User Interface:
Settings->Call Display->Display Contact Photo
Phone User Interface:
None phone_setting.called_party_info_display.enable 0 or 1 0
Description:
Enables or disables the IP phone to present the callee identity when it receives an incoming call.
Web User Interface:
Settings->Call Display->Display Called Party Information
Phone User Interface:
None phone_setting.call_info_display_method Integer from 0 to 4 0
Description:
Configures the call information display method when the IP phone receives an incoming call, dials an outgoing call or engages in a call.
0-Name+Number
1-Number+Name
2-Number
3-Name
4-Full Contact Info (display name<sip:[email protected]>)
64
Configuring Basic Features
Parameters Permitted Values
Web User Interface:
Settings->Call Display->Call Information Display Method
Phone User Interface:
None
Default
To configure call display features via web user interface:
1. Click on Settings->Call Display.
2. Select the desired value from the pull-down list of Display Contact Photo.
3. Select the desired value from the pull-down list of Display Called Party Information.
4. Select the desired value from the pull-down list of Call Information Display Method.
5. Click Confirm 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.
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 the pages returned by the web server. Both the HTTP and HTTPS port numbers are configurable.
Procedure
Web server type can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Specify the web access type,
HTTP port and HTTPS port.
Parameters: wui.http_enable network.port.http wui.https_enable
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Administrator’s Guide for SIP-T4X IP Phones
Local
Web User Interface
Phone User Interface network.port.https
Specify the web access type,
HTTP port and HTTPS port.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
Specify the web access type,
HTTP port and HTTPS port.
Details of Configuration Parameters:
Parameters Permitted Values Default wui.http_enable 0 or 1 1
Description:
Enables or disables the user to access web user interface of the IP phone using HTTP protocol.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->Web Server->HTTP
Phone User Interface:
Menu->Advanced (default password: admin)->Network->Webserver Type ->HTTP
Status network.port.http Integer from 1 to 65535 80
Description:
Configures the HTTP port for the user to access web user interface of the IP phone using the HTTP protocol.
The default HTTP port is 80.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->Web Server->HTTP Port (1~65535)
Phone User Interface:
Menu->Advanced (default password: admin)->Network->Webserver Type ->HTTP
66
Configuring Basic Features
Parameters
Port wui.https_enable
Permitted Values Default
0 or 1 1
Description:
Enables or disables the user to access web user interface of the IP phone using
HTTPS protocol.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->Web Server->HTTPS
Phone User Interface:
Menu->Advanced (default password: admin)->Network->Webserver Type ->HTTPS
Status network.port.https Integer from 1 to 65535 443
Description:
Configures the HTTPS port for the user to access web user interface of the IP phone using the HTTPS protocol.
The default HTTPS port is 443.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->Web Server->HTTPS Port (1~65535)
Phone User Interface:
Menu->Advanced (default password: admin)->Network->Webserver Type ->HTTPS
Port
To configure the web server type via web user interface:
1. Click on Network->Advanced.
2. In the Web Server block, select the desired value from the pull-down list of HTTP.
3. Enter the HTTP port in the HTTP Port (1~65535) field.
The default HTTP port is 80.
4. Select the desired value from the pull-down list of HTTPS.
5. Enter the HTTPS port in the HTTPS Port (1~65535) field.
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Administrator’s Guide for SIP-T4X IP Phones
The default HTTPS port is 443.
6. Click Confirm to accept the change.
A dialog box pops up to prompt that the settings will take effect after reboot.
7. Click OK to reboot the phone.
To configure the web server type via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->Webserver Type.
2. Press or , or the Switch soft key to select the desired value in the HTTP
Status field.
3. Enter the HTTP port in the HTTP Port field.
4. Press or , or the Switch soft key to select the desired value in the HTTPS
Status field.
5. Enter the HTTPS port in the HTTPS Port field.
6. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
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Some menu options are protected by two privilege levels, user and administrator, each with its own password. When logging into the web user interface, you need to enter the user name and password to access various menu options.
A user or an administrator can change the user password. The default user password is
“user”. For security reasons, the user or the administrator should change the default
Configuring Basic Features user password as soon as possible.
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.
Parameter: security.user_password
Change the user password of the
IP phone.
Navigate to: http://<phoneIPAddress>/servlet
?p=security&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default security.user_password String within 32 characters user
Description:
Configures the password of the user for web server access.
The IP phone uses “user” as the default user password.
The valid value format is username:new password.
Example: security.user_password = user:password123 means setting the password of user
(current user name is “user”) to password123.
Note: IP phones support ASCII characters 32-126(0x20-0x7E) in passwords. You can set the password to be empty via web user interface only.
Web User Interface:
Security->Password
Phone User Interface:
None
To change the user password via web user interface:
1. Click on Security->Password.
2. Select User from the pull-down list of User Type.
3. Enter a new password in the New Password and Confirm Password fields.
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Administrator’s Guide for SIP-T4X IP Phones
Valid characters are ASCII characters 32-126(0x20-0x7E) except 58(3A).
4. Click Confirm to accept the change.
Note
If logging into the web user interface of the IP phone with the user credential, the user needs to enter the current user password in the Old Password field.
Advanced menu options are strictly used by administrators. Users can configure them only if they have administrator privileges. The administrator password can only be changed by an administrator. The default administrator password is “admin”. For security reasons, the administrator should change the default administrator password as soon as possible.
Procedure
Administrator password can be changed using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Change the administrator password.
Parameter: security.user_password
Change the administrator password.
Navigate to: http://<phoneIPAddress>/servlet
?p=security&q=load
Change the administrator password.
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Configuring Basic Features
Details of the Configuration Parameter:
Parameter Permitted Values Default security.user_password String within 32 characters admin
Description:
Configures the password of the administrator for web server access.
The IP phone uses “admin” as the default administrator password.
Example: security.user_password = admin:password123 means setting the password of administrator (current user name is “admin”) to password123.
Note: IP phones support ASCII characters 32-126(0x20-0x7E) in passwords. You can set the password to be empty via web user interface only.
Web User Interface:
Security->Password
Phone User Interface:
Menu->Advanced (default password: admin) ->Set Password
To change the administrator password via web user interface:
1. Click on Security->Password.
2. Select admin from the pull-down list of User Type.
3. Enter the current administrator password in the Old Password field.
4. Enter a new administrator password in the New Password and Confirm Password fields.
Valid characters are ASCII characters 32-126(0x20-0x7E) except 58(3A).
5. Click Confirm to accept the change.
To change the administrator password via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Set Password.
2. Enter the current administrator password in the Current PWD field.
3. Enter a new administrator password in the New PWD field and Confirm PWD field.
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Administrator’s Guide for SIP-T4X IP Phones
4. Press the Save soft key to accept the change.
72
Phone lock is used to lock the IP phone to prevent it from unauthorized use. Phone lock is not applicable to SIP-T48G IP phones. 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 IP 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 phone lock key (if configured) when the IP phone is idle.
In addition to the above steps, you can configure IP phones to automatically lock the phone after a period of time.
Procedure
Phone lock can be configured using the configuration files or locally.
Configuration
File
Local
<y0000000000xx>.c
fg
Web User Interface
Configure the IP phone lock type.
Parameters: phone_setting.phone_lock.enable phone_setting.phone_lock.lock_key_type
Change the unlock password.
Parameter: phone_setting.phone_lock.unlock_pin
Configure the IP phone to automatically lock the phone after a time interval.
Parameter: phone_setting.phone_lock.lock_time_out
Assign a phone lock key.
Parameter: linekey.X.type/programablekey.X.type
Configure the phone lock type.
Change the unlock PIN.
Configure the IP phone to automatically lock the phone after a time interval.
Navigate to: http://<phoneIPAddress>/servlet?p=feat ures-phonelock&q=load
Assign a phone lock key.
Configuring Basic Features
Phone User
Interface
Navigate to: http://<phoneIPAddress>/servlet?p=dssk ey&model=1&q=load&linepage=1
Configure the phone lock type.
Change the unlock PIN.
Assign a phone lock key.
Details of Configuration Parameters:
Parameters Permitted Values Default phone_setting.phone_lock.enable 0 or 1 0
Description:
Enables or disables phone lock feature.
0-Disabled
1-Enabled
Note: It is not applicable to SIP-T48G IP phones.
Web User Interface:
Features->Phone Lock->Phone Lock Enable
Phone User Interface:
Menu->Advanced (default password: admin) ->Phone Settings->Phone Lock->Lock
Enable phone_setting.phone_lock.lock_key_type 0, 1 or 2 0
Description:
Configures the type of phone lock.
Menu Key: The Menu soft key is locked.
Function Keys: MESSAGE, REDIAL, HOLD, MUTE, TRANSFER, OK, X, navigation keys, soft keys and line keys are locked. (For SIP-T42G/T41P, HOLD key and TRANSFER key do not exist).
All Keys: All keys are locked, except the Volume, Headset, Speakerphone and digit keys.
0-All Keys
1-Function Keys
2-Menu Key
Note: It is not applicable to SIP-T48G IP phones.
Web User Interface:
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Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values Default
Features->Phone Lock->Phone Lock Type
Phone User Interface:
Menu->Advanced (default password: admin) ->Phone Settings->Phone Lock->Lock
Type phone_setting.phone_lock.unlock_pin characters within 15 digits
Description:
Configures the password for unlocking the phone.
Note: It is not applicable to SIP-T48G IP phones.
Web User Interface:
Features->Phone Lock->Phone Unlock PIN(0~15 Digit)
Phone User Interface:
Menu->Basic->Change PIN phone_setting.phone_lock.lock_time_out
Integer from 0 to
3600
123
0
Description:
Configures the interval (in seconds) to automatically lock the phone.
The default value is 0 (the phone is locked only by long pressing the pound key or pressing the phone lock key).
Note: It works only if the type of phone lock is preset. It is not applicable to SIP-T48G
IP phones.
Web User Interface:
Features->Phone Lock->Phone Lock Time Out(0~3600s)
Phone User Interface:
None
Phone Lock Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameter linekey.X.type/programablekey.X.type
Permitted Values
50
Default
Refer to the following content
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Configuring Basic Features
Parameter Permitted Values Default
Description:
Configures a DSS key as a phone lock key on the IP phone.
The digit 50 stands for the key type Phone Lock.
For line keys:
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Example: linekey.2.type = 50
Default:
For line keys:
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
For programable keys:
For
SIP-T46G
IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 (
Directory
).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61 (
Directory
).
When X=7, the default value is 51 (
Switch Account Up
).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=12, the default value is 0 ( NA ).
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Administrator’s Guide for SIP-T4X IP Phones
Parameter Permitted Values
When X=13, the default value is 0 (
NA
).
When X=14, the default value is 2 (
Forward
).
For
SIP-T42G/T41P
IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 (
Directory
).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61 (
Directory
).
When X=7, the default value is 51 ( Switch Account Up ).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 ( NA ).
When X=13, the default value is 0 ( NA ).
Note: It is not applicable to SIP-T48G IP phones.
Web User Interface:
DSSKey->Line Key X
/Programable Key
->Type
Phone User Interface:
Menu->Features->DSS Keys
->
Line Key X
->Type
Default
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 Phone Lock Enable.
3. Select the desired type from the pull-down list of Phone Lock Type.
4. Enter unlock password (numeric characters) in the Phone Unlock PIN (0~15 Digit) field.
5. Enter the desired time in the Phone Lock Time Out (0~3600s) field.
76
Configuring Basic Features
6. Click Confirm to accept the change.
To configure a phone lock key via web user interface:
1. Click on DSSKey->Line Key X.
2. In the desired DSS key field, select Phone Lock from the pull-down list of Type.
3. Click Confirm to accept the change.
To configure phone lock type via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Phone Settings->Phone Lock.
2. Press or , or the Switch soft key to select the desired value from the Lock
Enable field.
3. Press or , or the Switch soft key to select the desired type from the Lock
Type field.
4. Press the Save soft key to accept the change.
To change the unlock PIN via phone user interface:
1. Press Menu->Basic->Change 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 phone lock key via phone user interface:
1. Press Menu->Features->DSS Keys.
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Phone Lock from the Key Type field.
5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
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Administrator’s Guide for SIP-T4X IP Phones
6. Press the Save soft key to accept the change.
78
IP phones maintain a local clock and calendar. Time and date are displayed on the idle screen of the IP phone. Time and date are synced automatically from the NTP server by default. The NTP server can be obtained by DHCP or configured manually. If IP phones cannot obtain the time and date from the NTP server, you need to manually configure them. The time and date display can use one of several different formats.
Time Zone
A time zone is a region on Earth that has a uniform standard time. It is convenient for areas in close commercial or other communication to keep the same time. When configuring IP phones to obtain the time and date from the NTP server, you must set the time zone.
Daylight Saving Time
Daylight Saving Time (DST) is the practice of temporary advancing clocks during the summertime so that evenings have more daylight and mornings have less. Typically, clocks are adjusted forward one hour at the start of spring and backward in autumn.
Many countries have used the DST at various times, details vary by location. The DST can be adjusted automatically from the time zone configuration. Typically, there is no need to change this setting.
The following table lists available methods for configuring time and date:
Option
Time Zone
Time
Time Format
Date
Date Format
Methods of Configuration
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
Configuring Basic Features
Option
Daylight Saving Time
Methods of Configuration
Configuration Files
Web User Interface
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure NTP by DHCP priority feature and DHCP time features.
Parameters: local_time.manual_ntp_srv_prior local_time.dhcp_time
Configure the NTP server, time zone and DST.
Parameters: local_time.ntp_server1 local_time.ntp_server2 local_time.interval local_time.time_zone local_time.time_zone_name local_time.summer_time local_time.dst_time_type local_time.start_time local_time.end_time local_time.offset_time
Configure the time and date manually.
Parameter: local_time.manual_time_enable
Configure the time and date formats.
Parameters: local_time.time_format local_time.date_format
Configure NTP by DHCP priority feature.
Configure the NTP server, time zone and DST.
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Administrator’s Guide for SIP-T4X IP Phones
Phone User Interface
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.
Details of Configuration Parameters:
Parameters Permitted Values Default local_time.manual_ntp_srv_prior 0 or 1 0
Description:
Enables or disables the IP phone to use manually configured NTP server preferentially.
0-High (use the NTP server obtained by DHCP preferentially)
1-Low (use the NTP server configured manually preferentially)
Web User Interface:
Settings->Time & Date->NTP By DHCP Priority
Phone User Interface:
None local_time.dhcp_time 0 or 1 0
Description:
Enables or disables the IP phone to update time with the offset time obtained from the DHCP server.
0-Disabled
1-Enabled
Note: It is only available to offset from GMT 0.
Web User Interface:
Settings->Time & Date->DHCP Time
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Configuring Basic Features
Parameters
Phone User Interface:
Menu->Basic->Time & Date->DHCP Time
Permitted Values local_time.ntp_server1
Default
IP Address or Domain Name cn.pool.ntp.org
Description:
Configures the IP address or the domain name of the NTP server 1.
Example: local_time.ntp_server1 = 192.168.0.5
Web User Interface:
Settings->Time & Date->Primary Server
Phone User Interface:
Menu->Basic->Time & Date->General->SNTP Setting->NTP Server 1. local_time.ntp_server2 IP Address or Domain Name cn.pool.ntp.org
Description:
Configures the IP address or the domain name of the NTP server 2. If the NTP server 1 is not configured or cannot be accessed, the IP phone will request the time and date from the NTP server 2.
Example: local_time.ntp_server2 = 192.168.0.6
Web User Interface:
Settings->Time & Date->Secondary Server
Phone User Interface:
Menu->Basic->Time & Date->General->SNTP Setting->NTP Server 2. local_time.interval Integer from 15 to 86400 1000
Description:
Configures the interval (in seconds) to update time and date from the NTP server.
Example: local_time.interval = 1000
Web User Interface:
Settings->Time & Date->Synchronism (15~86400s)
Phone User Interface:
None
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Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values Default local_time.time_zone -11 to +14 +8
Description:
Configures the time zone.
For more available time zones, refer to Appendix B: Time Zones on page 511 .
Example: local_time.time_zone = +8
Web User Interface:
Settings->Time & Date->Time Zone
Phone User Interface:
Menu->Basic->Time & Date->General->SNTP Setting->Time Zone local_time.time_zone_name String within 32 characters China(Beijing)
Description:
Configures the time zone name.
The available time zone names depend on the time zone configured by the parameter “local_time.time_zone”. For more information on the available time zone
names for each time zone, refer to Appendix B: Time Zones on page 511 .
Note: It works only if the value of the parameter “local_time.summer_time” is set to 2
(Automatic).
Example: local_time.time_zone_name = China(Beijing)
Web User Interface:
Settings->Time & Date->Time Zone
Phone User Interface:
Menu->Basic->Time & Date->General->SNTP Setting->Time Zone local_time.summer_time 0, 1 or 2 2
Description:
Configures Daylight Saving Time (DST) feature.
0-Disabled
1-Enabled
2-Automatic
Web User Interface:
Settings->Time & Date->Daylight Saving Time
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Configuring Basic Features
Parameters Permitted Values
Phone User Interface:
Menu->Basic->Time & Date->General->SNTP Setting->Daylight Saving
Default local_time.dst_time_type 0 or 1 0
Description:
Configures the DST time type.
0-By Date
1-By Week
Note: It works only if the value of the parameter “local_time.summer_time” is set to 1
(Enabled).
Web User Interface:
Settings->Time & Date->Fixed Type
Phone User Interface:
None local_time.start_time Time 1/1/0
Description:
Configures the start time of the DST.
Value formats are:
Month/Day/Hour (for By Date)
Month/ Day of Week Last in Month/ Day of Week/ Hour of Day (for By Week)
If “local_time.dst_time_type” is set to 0 (By Date), use the mapping:
Month: 1=Jan, 2=Feb,…, 12=Dec
Day:1=the first day in a month,…, 31= the last day in a month
Hour:0=0am, 1=1am,…, 23=23pm
If “local_time.dst_time_type” is set to 1 (By Week), use the mapping:
Month: 1=Jan, 2=Feb,…, 12=Dec
Day of Week Last in 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=0am, 1=1am,…, 11=11pm
Note: It works only if the value of the parameter “local_time.summer_time” is set to 1
(Enabled).
Web User Interface:
For DST By Date:
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Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values Default
Settings->Time & Date->Start Date
For DST By Week:
Settings->Time & Date->DST Start Month/DST Start Day of Week/DST Start Day of
Week Last in Month/ Start Hour of Day
Phone User Interface:
None local_time.end_time Time 12/31/23
Description:
Configures the end time of the DST.
Value formats are:
Month/Day/Hour (for By Date)
Month/ Day of Week Last in Month/ Day of Week/ Hour of Day (for By Week)
If “local_time.dst_time_type” is set to 0 (By Date), use the mapping:
Month: 1=Jan, 2=Feb,…, 12=Dec
Day:1=the first day in a month,…, 31= the last day in a month
Hour:0=0am, 1=1am,…, 23=11pm
If “local_time.dst_time_type” is set to 1 (By Week), use the mapping:
Month: 1=Jan, 2=Feb,…, 12=Dec
Day of Week Last in 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=0am, 1=1am,…, 23=11pm
Note: It works only if the value of the parameter “local_time.summer_time” is set to 1
(Enabled).
Web User Interface:
For DST By Date:
Settings->Time & Date->End Date
For DST By Week:
Settings ->Time & Date->DST Stop Month/DST Stop Day of Week/DST Stop Day of
Week Last in Month/Stop Hour of Day local_time.offset_time Integer from -300 to 300 Blank
Description:
Configures the offset time (in minutes) of DST.
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Configuring Basic Features
Parameters Permitted Values Default
Note: It works only if the value of the parameter “local_time.summer_time” is set to 1
(Enabled).
Web User Interface:
Settings->Time & Date->Offset(minutes)
Phone User Interface:
None local_time.manual_time_enable 0 or 1 0
Description:
Configures the IP phone to obtain time from the NTP server or manual settings.
0-NTP
1-Manual
Web User Interface:
Settings->Time & Date->Manual Time
Phone User Interface:
Menu->Basic->Time & Date->General local_time.time_format 0 or 1 1
Description:
Configures the time format.
0-12 Hour
1-24 Hour
If it is set to 0 (12 Hour), the time will be displayed in 12-hour format with AM or PM specified.
If it is set to 1 (24 Hour), the time will be displayed in 24-hour format (eg., 2:00 PM displays as 14:00).
Web User Interface:
Settings->Time & Date->Time Format
Phone User Interface:
Menu->Basic->Time & Date->Time & Date Format->Time Format local_time.date_format 0, 1, 2, 3, 4, 5 or 6 0
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Administrator’s Guide for SIP-T4X IP Phones
Parameters Permitted Values Default
Description:
Configures the date format.
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
Web User Interface:
Settings->Time & Date->Date Format
Phone User Interface:
Menu->Basic->Time & Date->Time & Date Format->Date Format
To configure NTP by DHCP priority feature via web user interface:
1. Click on Settings->Time & Date.
2. Select the desired value from the pull-down list of NTP By DHCP Priority.
86
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.
Configuring Basic Features
4. Enter the domain names or IP addresses in the Primary Server and Secondary
Server fields respectively.
5. Enter the desired time interval in the Synchronism (15~86400s) field.
6. Select the desired value from the pull-down list of Daylight Saving Time.
If you select Enabled, do one of the following:
- 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.
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Administrator’s Guide for SIP-T4X IP Phones
Enter the desired time in the End Hour of Day field.
7. Enter the desired offset time in the Offset (minutes) field.
8. Click Confirm to accept the change.
To configure the time and date manually via web user interface:
1. Click on Settings->Time & Date.
2. Select Enabled from the pull-down list of Manual Time.
3. Enter the time and date in the corresponding fields.
88
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.
Configuring Basic Features
3. Select the desired value from the pull-down list of Date Format.
4. Click Confirm to accept the change.
To configure the NTP server and time zone via phone user interface:
1. Press Menu->Basic->Time & Date->General->SNTP Setting.
2. Press or , or the Switch soft key to select the time zone that applies to your area from the Time Zone field.
The default time zone is "+8 China(Beijing)".
3. Enter the domain names or IP addresses in the NTP Server 1 and NTP Server 2 fields respectively.
4. Press or, or the Switch soft key to select Automatic from the Daylight
Saving field.
5. Press the Save soft key to accept the change.
To configure the time and date manually via phone user interface:
1. Press Menu->Basic->Time & Date->General->Manual Setting.
2. Enter the specific time and date.
3. Press the Save soft key to accept the change.
To configure the time and date formats via phone user interface:
1. Press Menu->Basic->Time & Date->Time & Date Format.
2. Press or , or the Switch soft key to select the desired date format from the
Date Format field.
3. Press or , or the Switch soft key to select the desired time format (12 Hour or 24 Hour) from the Time Format field.
4. Press the Save soft key to accept the change.
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Administrator’s Guide for SIP-T4X IP Phones
IP phones support multiple languages. Languages used on the phone user interface and web user interface can be specified respectively as required.
The following table lists the languages supported by the phone user interface and the web user interface respectively.
Phone User Interface
English
Chinese Simplified
Chinese Traditional
French
German
Italian
Polish
Portuguese
Spanish
Turkish
Russian
Web User Interface
English
Chinese Simplified
Chinese Traditional
French
German
Italian
Polish
Portuguese
Spanish
Turkish
Russian
90
Languages available for selection depend on language packs currently loaded to the
IP phone. You can customize the translation of the existing language for the phone user interface or web user interface. You can also make new languages available for use on the phone user interface and web 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 the associated language packs for the phone user interface:
Available Language
English
Chinese Simplified
Chinese Traditional
French
German
Italian
Associated Language Pack for SIP-T48G/T46G/T42G/T41P
000.GUI.English.lang
001.GUI.Chinese_S.lang
002.GUI.Chinese_T.lang
003.GUI.French.lang
004.GUI.German.lang
005.GUI.Italian.lang
Configuring Basic Features
Available Language
Polish
Portuguese
Spanish
Turkish
Russian
Associated Language Pack for SIP-T48G/T46G/T42G/T41P
006.GUI.Polish.lang
007.GUI.Portuguese.lang
008.GUI.Spanish.lang
009.GUI.Turkish.lang
010.GUI.Russian.lang
When adding a new language pack for the phone user interface, the language pack must be formatted as “X.GUI.name.lang” (X starts from 011, “name” is replaced with the language name). If the language name is the same as the existing one, the existing language pack will be overridden by the new uploaded one. We recommend that the filename of the new language pack should not be the same as the existing one.
To customize a language file:
1. Open the desired language template file (e.g., 000.GUI.English.lang) using an
ASCII editor.
2. Modify the characters within the double quotation marks on the right of the equal sign.
Don’t modify the translation item on the left of the equal sign.
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The following shows a portion of the language pack “000.GUI.English.lang” for the phone user interface.
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The following table lists available languages, associated language packs and note language packs for the web user interface:
Available Language
English
Chinese_S
Chinese_T
French
German
Associated Language
Pack
1.English.js
2.Chinese_S.js
3.Chinese_T.js
4.French.js
5.German.js
Italian
Polish
Portuguese
Spanish
Turkish
Russian
6.Italian.js
7.Polish.js
8.Portuguese.js
9.Spanish.js
10.Turkish.js
11.Russian.js
Associated Note
Language Pack
1.English_note.xml
2.Chinese_S_note.xml
3.Chinese_T_note.xml
4.French_note.xml
5.German_note.xml
6.Italian_note.xml
7.Polish_note.xml
8.Portuguese_note.xml
9.Spanish_note.xml
10.Turkish_note.xml
11.Russian_note.xml
Configuring Basic Features
When adding a new language pack for the web user interface, the language pack must be formatted as “Y. name.js” (Y starts from 12, “name” is replaced with the language name). If the language name is the same as the existing one, the existing language file will be overridden by the new uploaded one. We recommend that the name of the new language file should not be the same as the existing languages.
To customize a language file:
1. Open the desired language template file (e.g., 1.English.js) using an ASCII editor.
2. Modify the characters within the double quotation marks on the right of the colon.
Don’t modify the translation item on the left of the colon.
The following shows a portion of the language pack “1.English.js” for the web user interface:
You can also customize the translation of the note language pack. The note information is integrated in the icon of the web user interface. The note language pack must be formatted as “Y.name_note.xml” (“Y” and “name” are associated with web language pack).
To customize a note language file:
1. Open the desired note language template file (e.g., 1.English_note.xml) using an
ASCII editor.
2. Modify the text of the note field.
Don't modify the name of the note field.
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The following shows a portion of the note language pack “1.English_note.xml” for the web user interface:
Note
If the characters in the custom language pack are not supported by the phone, the IP phone will display “?” instead.
The total file sizes of the custom language files must be within 20M (for SIP-T48G/T46G) or
100k (for SIP-T42G/T41P).
Procedure
Loading language pack can only be performed using the configuration files.
Configuration File <y0000000000xx>.cfg
Specify the access URL of the phone user interface language pack.
Parameter: gui_lang.url
Specify the access URL of the web user interface language pack.
Parameter: wui_lang.url
Specify the access URL of the note language pack of the web user interface.
Parameter: wui_lang_note.url
Delete customized language packs of the phone user interface.
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Configuring Basic Features
Parameter: gui_lang.delete
Delete customized language packs and note language packs of the web user interface.
Parameter: wui_lang.delete
Details of the Configuration Parameter:
Parameter Permitted Values Default gui_lang.url URL within 511 characters Blank
Description:
Configures the access URL of the language pack for the phone user interface.
Example:
The following example uses HTTP to download the language pack
“000.GUI.English.lang” from the provisioning server 192.168.10.25 to the phone user interface. gui_lang.url = http://192.168.10.25/000.GUI.English.lang
If you want to download multiple language packs to the SIP-T42G phone simultaneously, you can configure as following: gui_lang.url = http://192.168.10.25/000.GUI.English.lang gui_lang.url = http://192.168.10.25/010.GUI.Russian.lang
Web User Interface:
None
Phone User Interface:
None gui_lang.delete http://localhost/all or http://localhost/Y.GUI.nam
e.lang
Blank
Description:
Deletes customized language packs of the phone user interface.
Example:
Delete all customized language packs of the phone user interface. gui_lang.delete = http://localhost/all
Delete a customized language pack of the phone user interface (e.g.,
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Parameter Permitted Values
010.GUI.Russian.lang). gui_lang.delete = http://localhost/010.GUI.Russian.lang
Web User Interface:
None
Phone User Interface:
None wui_lang.url URL within 511 characters
Default
Blank
Description:
Configures the access URL of the language pack for the web user interface.
Example:
The following example uses HTTP to download the language pack “1.English.js” from the provisioning server 192.168.10.25 to the web user interface. wui_lang.url = http://192.168.10.25/1.English.js
If you want to download multiple language packs to the phone simultaneously, you can configure as following: wui_lang.url = http://192.168.10.25/1.English.js wui_lang.url = http://192.168.10.25/11.Russian.js
Web User Interface:
None
Phone User Interface:
None wui_lang_note.url
URL within 511 characters Blank
Description:
Configures the access URL of the language pack for web note.
Example:
The following example uses HTTP to download the language pack
“1.English_note.xml” from the provisioning server 192.168.10.25 to the web user interface. wui_lang_note.url = http://192.168.10.25/1.English_note.xml
If you want to download multiple language packs to the phone simultaneously, you can configure as following: wui_lang.url = http://192.168.10.25/1.English_note.xml wui_lang.url = http://192.168.10.25/11.Russian_note.xml
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Configuring Basic Features
Parameter
Web User Interface:
None
Phone User Interface:
None
Permitted Values Default wui_lang.delete http://localhost/all or http://localhost/Y.
name
.js
Blank
Description:
Deletes customized language packs and note language packs of the web user interface.
Example:
Delete all customized language packs and note language packs of the web user interface. web_lang.delete = http://localhost/all
Delete a customized language pack (e.g., 11.Russian.js) of the web user interface. gui_lang.delete = http://localhost/11.Russian.js
The corresponding note language pack (e.g., 11.Russian_note.xml) will also be deleted.
Web User Interface:
None
Phone User Interface:
None
The default language used on the phone user interface is English. You can specify the languages for the phone user interface and web user interface respectively.
Procedure
Specify the language for the web user interface or the phone user interface using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Specify the languages for the phone user interface and the web user interface.
Parameters: lang.gui
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Local
Web User Interface
Phone User Interface lang.wui
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.
Details of Configuration Parameters:
Parameters Permitted Values Default lang.gui Refer to the following content English
Description:
Configures the language used on the phone user interface.
Permitted Values:
English, Chinese_S, Chinese_T, French, German, Turkish, Italian, Polish, Spanish,
Portuguese, Russian or the custom language name.
Example: lang.gui = English
Web User Interface:
None
Phone User Interface:
Menu->Basic->Language lang.wui Refer to the following content Blank
Description:
Configures the language used on the web user interface.
Example: lang.wui = French
Permitted Values:
English, Chinese_S, Chinese_T, French, German, Italian, Polish, Spanish, Turkish,
Russian, Portuguese or the custom language name.
Web User Interface:
Settings->Preference->Language
Phone User Interface:
None
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Configuring Basic Features
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.
3. Click Confirm to accept the change.
To specify the language for the phone user interface via phone user interface:
1. Press Menu->Basic->Language.
2. Press or to select the desired language.
3. Press the Save soft key to accept the change.
Input method customization allows users to customize the existing input method on IP phones. You can first customize the Yealink-supplied input method file “ime.txt”, and then download it to the IP phone. IP phones support 5 input methods: 2aB, abc, Abc, 123,
ABC.
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The following shows a portion of the input method file “ime.txt”:
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Configuring Basic Features
You can add new characters or adjust the character order of the existing input method.
The following show an example of adding the Russian characters for the input method
“abc”.
Note
When adding new characters for the existing input method, ensure that the added characters are supported by IP phones.
The IP phones can only recognize the input method files uploaded using Unicode encoding.
Do not rename the input mode filename.
In addition to customizing the input method file, you can also specify the default input method for the IP phone when editing or searching for contacts.
Procedure
Specify the access URL of the custom input method file and the default input methods using the configuration files.
Configuration File <y0000000000xx>.cfg
Specify the access URL of the custom input method file.
Parameter: gui_input_method.url
Specify the default input method when editing contacts.
Parameter: directory.edit_default_input_meth
od
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Specify the default input method when searching for contacts.
Parameter: directory.search_default_input_m
ethod
Details of Configuration Parameters:
Parameters Permitted Values Default gui_input_method.url URL within 511 characters Blank
Description:
Configures the access URL of the custom input method file.
Example:
The following example uses HTTP to download the custom input method file (ime.txt) from the provisioning server 192.168.10.25. gui_input_method.url = http://192.168.10.25/ime.txt
Web User Interface:
None
Phone User Interface:
None directory.edit_default_input_method Abc, 2aB, 123, abc or ABC Abc
Description:
Specify the default input method for editing the contact information.
Example: directory.edit_default_input_method = abc
Web User Interface:
None
Phone User Interface:
None directory.search_default_input_method
Abc, 2aB, 123, abc or ABC Abc
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Configuring Basic Features
Default Parameters Permitted Values
Description:
Specify the default input method for searching the contacts.
Example: directory.search_default_input_method = abc
Web User Interface:
None
Phone User Interface:
None
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.
The logo file format must be *.dob, and the resolution of SIP-T42G/T41P IP phones is
192*64 graphic. Logo is not applicable to SIP-T48G and SIP-T46G IP phones. These two IP phone models use wallpaper instead.
Note
Before uploading your custom logo to IP phones, ensure the logo file is correctly formatted. For more information on customizing a logo file, refer to
Yealink_SIP-T2_Series_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 and specify the access URL of the custom logo file.
Parameters: phone_setting.lcd_logo.mode lcd_logo.url
Configure the logo shown on the idle screen.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
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Details of Configuration Parameters:
Parameters Permitted Values Default phone_setting.lcd_logo.mode 0, 1 or 2 0
Description:
Configures the logo mode of the LCD screen.
0-Disabled
1-System logo
2-Custom logo
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 IP phone).
Note: It is not applicable to SIP-T48G/T46G IP phones.
Web User Interface:
Features->General Information->Use Logo
Phone User Interface:
None lcd_logo.url URL within 511 characters Blank
Description:
Configures the access URL of the custom logo file.
Example:
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
Note: It is not applicable to SIP-T48G/T46G IP phones.
Web User Interface:
Features->General Information->Upload Logo
Phone User Interface:
None
To configure a custom logo via web user interface:
1. Click on Features->General Information.
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2. Select Custom logo from the pull-down list of Use Logo.
Configuring Basic Features
3. Click Browse to select the logo file from your local system.
4. Click Upload to upload the file.
5. Click Confirm to accept the change.
The custom logo screen and the idle screen are displayed alternately.
Softkey layout is used to customize the soft keys at the bottom of the LCD screen to best meet users’ requirements. In addition to specifying which soft keys to display, you can determine their display order. It can be configured based on call states.
You can configure the softkey layout using the softkey layout templates for different call states. For more information on how to configure a softkey layout template, refer to
Softkey Layout Template on page 484 .
The following table lists the soft keys available for IP phones in different states:
CallFailed
Call State Default Soft Key
NewCall
Empty
Empty
Empty
Optional Soft Key
Empty
Switch
Cancel
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CallIn
Connecting
Call State
Connecting
SemiAttendTrans
Dialing (not applicable to SIP-T48G)
Default Soft Key
Answer
Forward
Silence
Reject
Empty
Empty
Empty
Cancel
Transfer
Empty
Empty
Cancel
Send
IME
Delete
Cancel
Optional Soft Key
Empty
Switch
Empty
Switch
Empty
Switch
Empty
History
Switch
Line
Directory
GPickup
DPickup
Retrieve
Empty
Switch
CC
RingBack
Talking
RingBack
Empty
Empty
Empty
Cancel
SemiAttendTransBack
Transfer
Empty
Empty
Cancel
Talk
Transfer
Hold
Conference
Cancel
Empty
Switch
CC
Empty
Mute
SWAP
NewCall
Switch
Answer
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Configuring Basic Features
Call State
Hold
Held
PreTrans (not applicable to
SIP-T48G)
Conferenced
Default Soft Key
Transfer
Resume
NewCall
Cancel
Empty
Empty
Empty
Cancel
Transfer
IME
Delete
Cancel
Empty
Hold
Split
Cancel
Empty
Switch
Answer
Reject
NewCall
Empty
Directory
Switch
Send
Empty
Switch
Answer
Reject
Mute
Manager
Optional Soft Key
Reject
PriHold
Park
GPark
Empty
Switch
Answer
Reject
Procedure
Softkey layout can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Specify the access URL of the softkey layout template.
Parameters: phone_setting.custom_softkey_en
able custom_softkey_call_failed.url custom_softkey_call_in.url custom_softkey_connecting.url custom_softkey_dialing.url
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Local Web User Interface custom_softkey_ring_back.url custom_softkey_talking.url
Configure the softkey layout.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-softkey&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default phone_setting.custom_softkey_enable 0 or 1 0
Description:
Enables or disables custom soft keys layout feature.
0-Disabled
1-Enabled
Web User Interface:
Settings->Softkey Layout->Custom Softkey
Phone User Interface:
None custom_softkey_call_failed.url URL within 511 characters Blank
Description:
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Call Failed state.
Example:
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.xml
Web User Interface:
None
Phone User Interface:
None custom_softkey_call_in.url URL within 511 characters Blank
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Configuring Basic Features
Parameters Permitted Values Default
Description:
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Call In state.
Example:
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
Web User Interface:
None
Phone User Interface:
None custom_softkey_connecting.url URL within 511 characters Blank
Description:
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Connecting state.
Example:
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
Web User Interface:
None
Phone User Interface:
None custom_softkey_dialing.url URL within 511 characters Blank
Description:
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Dialing state.
Example:
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. custom_softkey_dialing.url = http://10.2.8.16:8080/XMLfiles/Dialing.xml
Web User Interface:
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Parameters
None
Phone User Interface:
None custom_softkey_ring_back.url
Permitted Values
URL within 511 characters
Default
Blank
Description:
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the RingBack state.
Example:
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.xml
Web User Interface:
None
Phone User Interface:
None custom_softkey_talking.url URL within 511 characters Blank
Description:
Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Talking state.
Example:
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
Web User Interface:
None
Phone User Interface:
None
To configure softkey layout via web user interface:
1. Click on Settings->Softkey Layout.
2. Select the desired value from the pull-down list of Custom Softkey.
3. Select the desired state from the pull-down list of Call States.
4. Select the desired soft key from the Unselected Softkeys column and click .
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Configuring Basic Features
The selected soft key appears in the Selected Softkeys column. If more than four soft keys are selected, a More soft key appears on the LCD screen, and the selected soft keys are displayed in two pages.
5. Repeat the step 4 to add more soft keys to the Selected Softkeys column.
6. Click to remove the soft key from the Selected Softkeys column.
7. Click or to adjust the display order of the soft key.
8. Click Confirm to accept the change.
Key as send allows assigning the pound key or asterisk key as a send key. Send sound allows the IP phone to play a key tone when a user presses the send key. Key tone allows the IP phone to play a key tone when a user presses any key. Send sound works only if Key tone is enabled.
Procedure
Key as send can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Configure a send key.
Parameter: features.key_as_send
Configure a send sound.
Parameter: features.send_key_tone
Configure a key tone.
Parameter: features.key_tone
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Local
Web User Interface
Phone User Interface
Configure a send key.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure a key tone and send tone.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-audio&q=load
Configure the send key.
Configure a key tone.
Details of Configuration Parameters:
Parameters Permitted Values Default features.key_as_send 0, 1 or 2 1
Description:
Configures the "#" or "*" key as the send key.
0-Disabled
1-# key
2-* key
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.
Note: The old parameter “features.pound_key.mode” is also applicable to IP phones.
Web User Interface:
Features->General Information->Key As Send
Phone User Interface:
Menu->Features->Others->General->Key As Send features.key_tone 0 or 1 1
Description:
Enables or disables the IP phone to play a tone when a user presses a key on your phone phone.
0-Disabled
1-Enabled
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Configuring Basic Features
Parameters Permitted Values Default
If it is set to 1 (Enabled), the IP phone will play a tone when a user presses a key on your phone phone.
Web User Interface:
Features->Audio->Key Tone
Phone User Interface:
Menu->Basic->Sound->Key Tone features.send_key_tone 0 or 1 1
Description:
Enables or disables the IP phone to play a tone when a user presses a send key.
0-Disabled
1-Enabled
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 value of the parameter “features.key_tone” is set to 1
(Enabled).
Web User Interface:
Features->Audio->Send Sound
Phone User Interface:
None
To configure a send key 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 Key As Send.
3. Click Confirm to accept the change.
To configure a key tone and send tone via web user interface:
1. Click on Features->Audio.
2. Select the desired value from the pull-down list of Key Tone.
3. Select the desired value from the pull-down list of Send Sound.
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4. Click Confirm to accept the change.
To configure a send key via phone user interface:
1. Press Menu->Features->Others->General.
2. Press or , or the Switch soft key to select # or * from the Key as Send field, or select Disabled to disable this feature.
3. Press the Save soft key to accept the change.
Configuring Basic Features
To configure a key tone via web user interface:
1. Press Menu->Basic->Sound->Key Tone.
2. Press or , or the Switch soft key to select the desired type from the Key Tone field.
3. Press the Save soft key to accept the change.
Note
Send tone works only if key tone is enabled.
Key tone is enabled by default.
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.
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”.
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[]
()
$
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”.
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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 482 .
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.
Parameters: dialplan.replace.prefix.X dialplan.replace.replace.X dialplan.replace.line_id.X
Configure the access URL of the replace rule template.
Parameter: dialplan_replace_rule.url
Create the replace rule for the IP phone.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-dialplan&q=load
Configuring Basic Features
Details of Configuration Parameters:
Parameters Permitted Values Default dialplan.replace.prefix.X
(X ranges from 1 to 100)
String within 32 characters
Description:
Configures the entered number to be replaced.
Example: dialplan.replace.prefix.1 = 00
Web User Interface:
Settings->Dial Plan->Replace Rule->Prefix
Phone User Interface:
None dialplan.replace.replace.X
(X ranges from 1 to 100)
String within 32 characters
Blank
Blank
Description:
Configures the alternate number to replace the entered number.
Example: dialplan.replace.replace.1 = 123456
Web User Interface:
Settings->Dial Plan->Replace Rule->Replace
Phone User Interface:
None dialplan.replace.line_id.X
(X ranges from 1 to 100)
Integer Blank (for all lines)
Description:
Configures the desired line to apply the replace rule. The digit 0 stands for all lines.
If it is left blank, the replace rule will apply to all lines on the IP phone.
0 to 16 (for SIP-T48G/T46G)
0 to 12 (for SIP-T42G)
0 to 6 (for SIP-T41P)
Example: dialplan.replace.line_id.1 = 1
Note: Multiple line IDs are separated by commas.
Web User Interface:
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Parameters Permitted Values
Settings->Dial Plan->Replace Rule->Account
Phone User Interface:
None dialplan_replace_rule.url URL within 511 characters
Default
Blank
Description:
Configures the access URL of the replace rule template file.
Example: dialplan_replace_rule.url = http://192.168.10.25/dialplan.xml
Web User Interface:
None
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 the field blank or enter 0, the replace rule will apply to all accounts on the IP phone.
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5. Click Add to add the replace rule.
Configuring Basic Features
Dial-now is a string used to match the numbers entered by the user. When entered numbers match the predefined dial-now rule, IP phones will automatically dial out the numbers without pressing the send key. IP phones support up to 100 dial-now rules, which can be created either one by one or in batch using a dial-now rule template. For
more information on the dial-now template, refer to Dial-now Template on page 483 .
Delay Time for Dial-now Rule
IP phones will automatically dial out the entered number, which matches the dial-now rule, after a specified period of time.
Procedure
Dial-now rule can be created using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Create the dial-now rule for the IP phone.
Parameters: dialplan.dialnow.rule.X dialplan.dialnow.line_id.X
Configure the delay time for the dial-now rule and the access URL of the dial-now template.
Parameters: phone_setting.dialnow_delay dialplan_dialnow.url
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
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Details of Configuration Parameters:
Parameters Permitted Values Default dialplan.dialnow.rule.X
(X ranges from 1 to 100)
String within 511 characters Blank
Description:
Configures the dial-now rule (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.
Example: dialplan.dialnow.rule.1 = 1234
Web User Interface:
Settings->Dial Plan->Dial-now->Rule
Phone User Interface:
None dialplan.dialnow.line_id.X
(X ranges from 1 to 100)
Integer Blank (for all lines)
Description:
Configures the desired line to apply the dial-now rule. The digit 0 stands for all lines.
If it is left blank, the dial-now rule will apply to all lines on the IP phone.
0 to 16 (for SIP-T48G/T46G)
0 to 12 (for SIP-T42G)
0 to 6 (for SIP-T41P)
Example: dialplan.dialnow.line_id.1 = 1
Note: Multiple line IDs are separated by commas.
Web User Interface:
Settings->Dial Plan->Dial-now->Account
Phone User Interface:
None phone_setting.dialnow_delay Integer from 1 to 14 1
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Configuring Basic Features
Parameters Permitted Values Default
Description:
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.
Web User Interface:
Features->General Information->Time-Out for Dial-Now Rule
Phone User Interface:
None dialplan_dialnow.url URL within 511 characters Blank
Description:
Configures the access URL of the dial-now rule template file.
Example: dialplan_dialnow.url = http://192.168.10.25/dialnow.xml
Web User Interface:
None
Phone User Interface:
None
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.
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If you leave the field blank or enter 0, the dial-now rule will apply to all accounts on the
IP phone.
4. Click Add to add the dial-now rule.
To configure the delay time for the dial-now rule via web user interface:
1. Click on Features->General Information.
2. Enter the desired time within 1-14 (in seconds) in the Time-Out for Dial-Now Rule field.
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3. Click Confirm to accept the change.
Configuring Basic Features
Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country. When the entered numbers match the predefined area code rule, the IP phone will automatically add the area code before the numbers when dialing out them. IP phones only support one area code rule.
Procedure
Area code rule can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Create the area code rule and specify the maximum and minimum lengths of the entered numbers.
Parameters: dialplan.area_code.code dialplan.area_code.min_len dialplan.area_code.max_len dialplan.area_code.line_id
Create the area code rule and specify the maximum and minimum lengths of entered numbers.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-areacode&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default dialplan.area_code.code String within 16 characters Blank
Description:
Configures the area code to be added before the entered numbers when dialing out.
Note: The length of the entered number must be between the minimum length configured by the parameter “dialplan.area_code.min_len” and the maximum length configured by the parameter “dialplan.area_code. max_len”.
Example: dialplan.area_code.code = 010
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Parameters
Web User Interface:
Settings->Dial Plan->Area Code->Code
Phone User Interface:
None
Permitted Values dialplan.area_code.min_len Integer from 1 to 15
Description:
Configures the minimum length of the entered numbers.
Web User Interface:
Settings->Dial Plan->Area Code->Min Length (1-15)
Phone User Interface:
None dialplan.area_code.max_len Integer from 1 to 15
Default
1
15
Description:
Configures the maximum length of the entered numbers.
Note: The value must be larger than the minimum length.
Web User Interface:
Settings->Dial Plan->Area Code->Max Length (1-15)
Phone User Interface:
None dialplan.area_code.line_id Integer Blank (for all lines)
Description:
Configures the desired line to apply the area code rule. The digit 0 stands for all lines. If it is left blank, the area code rule will apply to all lines on the IP phone.
0 to 16 (for SIP-T48G/T46G)
0 to 12 (for SIP-T42G)
0 to 6 (for SIP-T41P)
Example: dialplan.area_code.line_id = 1,2
Note: Multiple line IDs are separated by commas.
Web User Interface:
Settings->Dial Plan->Area Code->Account
Phone User Interface:
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Configuring Basic Features
Parameters Permitted Values Default
None
To configure an area code rule via web user interface:
1. Click on Settings->Dial Plan->Area Code.
2. Enter desired values in the Code, Min Length (1-15) and Max Length (1-15) fields.
3. Enter the desired line ID in the Account field or leave it blank.
If you leave the field blank or enter 0, the area code rule will apply to all accounts on the
IP phone.
4. Click Confirm to accept the change.
Block out rule prevents users from dialing out specific numbers. When the entered numbers match the predefined block out rule, the LCD screen prompts “Forbidden
Number”. IP phones support up to 10 block out rules.
Procedure
Block out rule can be created using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Create the block out rule for the
IP phone.
Parameters: dialplan.block_out.number.X dialplan.block_out.line_id.X
Create the block out rule for the desired line.
Navigate to: http://<phoneIPAddress>/servlet
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?p=settings-blackout&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default dialplan.block_out.number.X
(X ranges from 1 to 10)
String within 32 characters
Description:
Configures the block out numbers.
Example: dialplan.block_out.number.1 = 6666
Web User Interface:
Settings->Dial Plan->Block Out->BlockOut NumberX
Phone User Interface:
None dialplan.block_out.line_id.X
(X ranges from 1 to 10)
Integer
Blank
Blank (for all lines)
Description:
Configures the desired line to apply the block out rule. The digit 0 stands for all lines.
If it is left blank, the block out rule will apply to all lines on the IP phone
.
0 to 16 (for SIP-T48G/T46G)
0 to 12 (for SIP-T42G)
0 to 6 (for SIP-T41P)
Example: dialplan.block_out.line_id.1 = 1
Note: Multiple line IDs are separated by commas.
Web User Interface:
Settings->Dial Plan->Block Out->Account
Phone User Interface:
None
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.
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Configuring Basic Features
If you leave the field blank or enter 0, the block out rule will apply to all accounts on the
IP phone.
4. Click Confirm to add the block out rule.
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.
Parameter: features.hotline_number
Specify the time (in seconds) the
IP phone waits to automatically dial out the hotline number.
Parameter: features.hotline_delay
Configure the hotline number.
Specify the time (in seconds) the
IP phone waits to automatically dial out the hotline number.
Navigate to: http://<phoneIPAddress>/servlet
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Phone User Interface
?p=features-general&q=load
Configure the hotline number.
Specify the time (in seconds) the
IP phone waits to automatically dial out the hotline number.
Details of Configuration Parameters:
Parameter Permitted Values Default features.hotline_number String within 32 characters Blank
Description:
Configures the hotline 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.
Example: features.hotline_number = 3601
Web User Interface:
Features->General Information->Hotline Number
Phone User Interface:
Menu->Features->Others->Hot Line->Number features.hotline_delay Integer from 0 to 10 4
Description:
Configures the waiting time (in seconds) for the IP phone to automatically dial 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 designated seconds before dialing out the predefined hotline number when you lift the handset, press the speakerphone key or press the line key.
Web User Interface:
Features->General Information->Hotline Delay (0~10s)
Phone User Interface:
Menu->Features->Others->Hot Line->HotLine Delay
To configure hotline via web user interface:
1. Click on Features->General Information.
2. Enter the hotline number in the Hotline Number field.
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3. Enter the delay time in the Hotline Delay (0~10s) field.
Configuring Basic Features
4. Click Confirm to accept the change.
To configure hotline via phone user interface:
1. Press Menu->Features->Others->Hot Line.
2. Enter the hotline number in the Number field.
3. Enter the delay time in the HotLine Delay field.
4. Press the Save soft key to accept the change.
For security reasons, IP phones support off hook hot line dialing feature, which allows the phone to first dial out the pre-configured number when the user presses the speakerphone key or desired line key, dials out a call or off hook the phone using the account with this feature enabled. The SIP server may then prompt the user to enter an activation code for call service. Only if the user enters a valid activation code, the IP phone will be permitted to place the call.
Off hook hot line dialing feature is configurable on a per-line basis and depends on support from a SIP server.
Note
Off hook hot line dialing feature limits the call-out permission of this account and disables the hotline feature. For example, when the phone goes off
hook using the account with this feature enabled, the configured hotline number will not be dialed out automatically.
The server actions may vary from different servers.
This feature is also applicable to the IP call and intercom call.
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Procedure
Off hook hot line dialing can be configured using the configuration files.
Configuration File <y0000000000xx>.cfg
Configure off hook hot line dialing feature.
Parameter: account.X.auto_dial_enable
Specify the number that the phone first dials out.
Parameter: account.X.auto_dial_num
Details of Configuration Parameters:
Parameter Permitted Values Default account.X.auto_dial_enable 0 or 1 0
Description:
Enables or disables the IP phone to first dial out a pre-configured number when a user presses the speakerphone key or desired line key, dials out a call or off hook the phone using account X.
0-Disabled
1-Enabled
If it is set to 1(Enabled), the phone will first dial out the pre-configured number
(configured by the parameter “account.X.auto_dial_num”) when a user presses the speakerphone key or desired line key, dials out a call or off hook the phone using account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
None
Phone User Interface:
None account.X.auto_dial_num String within 32 characters Blank
Description:
Configures the number that the IP phone first dials out when a user presses the speakerphone key or desired line key, dials out a call or off hook the phone using
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Configuring Basic Features
Parameter Permitted Values Default account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Note: It works only if the value of the parameter “account.X.auto_dial_enable” is set to 1 (Enabled).
Web User Interface:
None
Phone User Interface:
None
Directory provides easy access to frequently used lists. The lists can be Local Directory,
History, Remote Phone Book and LDAP. The desired list(s) can be added to Directory using a directory file. For more information on the directory file, refer to
Procedure
Directory can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Specify the access URL of the
Directory file.
Parameter: directory_setting.url
Configure the Directory.
Navigate to: http://<phoneIPAddress>/servlet
?p=contacts-favorite&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default directory_setting.url URL within 511 characters Blank
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Parameter Permitted Values Default
Description:
Configures the access URL of the directory template.
Example: directory_setting.url = http://192.168.1.20/favorite_setting.xml
Web User Interface:
Directory->Setting->Directory
Phone User Interface:
None
To configure the directory via web user interface:
1. Click on Directory->Setting.
2. In the Directory block, select the desired list from the Disabled column and then click .
The selected list appears in the Enabled column.
3. Repeat the step 2 to add more lists to the Enabled column.
4. To remove a list from the Enabled column, select the desired list and then click .
5. To adjust the display order of enabled lists, select the desired list and then click or .
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6. Click Confirm to accept the change.
The IP phone LCD screen will display the enabled list(s) in the adjusted order.
Configuring Basic Features
Search source list in dialing allows the IP phone to automatically search entries from the search source list based on the entered string, and display results on the pre-dialing screen. The search source list can be Local Directory, History, Remote Phone Book and
LDAP. You can configure the search source list in dialing using a super search file. For
more information on the super search template, refer to Super Search Template on page 486 .
Procedure
Search source list can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Specify the access URL of the super search file.
Parameter: super_search.url
Configure the search source list in dialing.
Navigate to: http://<phoneIPAddress>/servlet
?p=contacts-favorite&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default super_search.url URL within 511 characters Blank
Description:
Configures the access URL of the super search template.
Web User Interface:
Directory->Setting->Search Source List In Dialing
Phone User Interface:
None
To configure search source list in dialing via web user interface:
1. Click on Directory->Setting.
2. In the Search Source List In Dialing block, select the desired list from the Disabled column and click .
The selected list appears in the Enabled column.
3. Repeat step 2 to add more lists to the Enabled column.
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4. To remove a list from the Enabled column, select the desired list and then click .
5. To adjust the display order of the enabled list, select the desired list, and click or .
6. Click Confirm to accept the change.
The pre-dialing screen displays the search results in the adjusted order.
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Call log contains call information such as remote party identification, time and date, and call duration. IP phones maintain a local call log. Call log consists of four lists:
Missed calls, Placed calls, Received calls and Forwarded calls. Each call log list supports up to 100 entries. To store call information, you must enable the save call log feature in advance.
Procedure
Call log can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure the call log.
Parameter: features.save_call_history
Configure the call log.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure the call log.
Configuring Basic Features
Details of the Configuration Parameter:
Parameter Permitted Values Default features.save_call_history 0 or 1 1
Description:
Enables or disables the IP phone to save call log.
0-Disabled
1-Enabled
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.
Web User Interface:
Features->General Information->Save Call Log
Phone User Interface:
Menu->Features->Others->General->History Record
To configure the call log via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Save Call Log.
3. Click Confirm to accept the change.
To configure the call log via phone user interface:
1. Press Menu->Features->Others->General.
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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.
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Missed call log allows IP phones to display the number of the missed calls with an indicator icon on the idle screen, and to log the missed calls in the Missed Calls list when 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 the missed call log feature.
Parameter: account.X.missed_calllog
Configure the missed call log feature.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-basic&q=load&acc
=0
Details of the Configuration Parameter:
Parameter Permitted Values Default account.X.missed_calllog 0 or 1 1
Description:
Enables or disables the IP phone to record missed calls for account X.
0-Disabled
1-Enabled
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.
Configuring Basic Features
Parameter
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Basic->Missed Call Log
Phone User Interface:
None
Permitted Values
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.
4. Select the desired value from the pull-down list of Missed Call Log.
Default
5. Click Confirm to accept the change.
The IP phone maintains a local directory. The local directory can store up to 1000 contacts and 48 groups (including the default groups: Company, Family and Friend).
When adding a contact to the local directory, in addition to name and phone numbers, you can also specify the account, ring tone and group for the contact. Contacts and groups can be added either one by one or in batch using a local contact file. Yealink IP phones support both *.xml and *.csv format contact files. For more information on how
to customize a contact file (*.xml), refer to Local Contact File on page 488 .
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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 (*.xml).
Parameter: local_contact.data.url
Add a new group and a contact to the local directory.
To import or export contact file.
Navigate to: http://<phoneIPAddress>/servlet
?p=contactsbasic&q=load&num
=1&group=
Add a new group and a contact to the local directory.
Details of the Configuration Parameter:
Parameter Permitted Values Default local_contact.data.url URL within 511 characters
Description:
Configures the access URL of the local contact file (*.xml).
Example: local_contact.data.url = http://192.168.10.25/contact.xml
Web User Interface:
Directory->Local Directory->Import Local Directory File
Phone User Interface:
None
To add a new group to the local directory via web user interface:
1. Click on Directory->Local Directory.
2. In the Group Setting block, enter the new group name in the Group field.
Blank
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Configuring Basic Features
3. Select the desired group ring tone from the pull-down list of Ring.
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. Enter the name and the office, mobile or other numbers in the corresponding fields.
3. Select the desired ring tone from the pull-down list of Ring Tone.
4. Select the desired group from the pull-down list of Group.
5. Select the desired account from the pull-down list of Account.
6. Select the desired photo from the pull-down list of Photo.
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It is not applicable to SIP-T42G and SIP-T41P IP phones.
7. Click Add to add the contact.
To import an XML contact list file via web user interface:
1. Click on Directory->Local Directory.
2. Click Browse to locate a contact list file (the file format must be *.xml) from your local system.
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3. Click Import XML to import the contact list.
Configuring Basic Features
The web user interface prompts "The original contact will be covered, Continue?".
4. Click OK to complete importing the contact list.
To import a CSV contact list file via web user interface:
1. Click on Directory->Local Directory.
2. Click Browse to locate a contact list file (the file format must be *.csv) from your local system.
3. (Optional.) Check the Show Title checkbox.
It will prevent importing the title of the contact information which is located in the first line of the CSV file.
4. Click Import CSV to import the contact list.
5. (Optional.) Mark the On radio box in the Delete Old Contacts field.
It will delete all existing contacts while importing the contact list.
6. Select the contact information you want to import into the local directory from the pull-down list of Index.
At least one row information should be selected to be imported into the local directory.
7. Click Import to complete importing the contact list.
To export a contact list via web user interface:
1. Click on Directory->Local Directory.
2. Click Export XML (or Export CSV).
3. Click Save to save the contact list to your local system.
To add a group to the local directory via phone user interface:
1. Press Menu->Directory->Local Directory.
2. Press the Add Group soft key.
3. Enter the desired group name in the Name field.
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4. Press or to select the desired group ring tone from the Ring Tones field.
5. Press the Save soft key to accept the change or the Back soft key to cancel.
To add a contact to the local directory via phone user interface:
1. Press Menu->Directory->Local Directory.
2. Select the desired contact group and then press the Enter soft key.
3. Press the Add soft key.
4. Enter the name and the office, mobile or other numbers in the corresponding fields.
5. Press or , or the Switch soft key to select the desired account from the
Account field.
If Auto is selected, the IP phone will use the first available account when placing calls to the contact from the local directory.
6. Press or , or the Switch soft key to select the desired ring tone from the Ring field.
7. Press or , or the Switch soft key to select the desired photo from the Photo field.
8. Press the Save soft key to accept the change.
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.
Parameters: phone_setting.predial_autodial phone_setting.inter_digit_time
Configure live dialpad.
Navigate to: http://<phoneIPAddress>/servlet
?p=settings-preference&q=load
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Configuring Basic Features
Details of Configuration Parameters:
Parameters Permitted Values Default phone_setting.predial_autodial 0 or 1 0
Description:
Enables or disables live dialpad feature.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the IP phone will automatically dial out the entered phone number in the pre-dialing screen without pressing a send key.
Web User Interface:
Settings->Preference->Live Dialpad
Phone User Interface:
None phone_setting.inter_digit_time Integer from 1 to 14 4
Description:
Configures the time (in seconds) for the IP phone to automatically dial out the entered digits without pressing a send key.
Note: It works only if the value of the parameter “phone_setting.predial_autodial” is set to 1 (Enabled).
Web User Interface:
Settings->Preference->Inter Digit Time (1~14s)
Phone User Interface:
None
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.
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3. Enter the desired delay time in the Inter Digit Time (1~14s) field.
4. Click Confirm to accept the change.
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Call waiting allows IP phones to receive a new incoming 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 IP phone to play a short tone, to remind the user audibly of a new incoming call during conversation. Call waiting tone works only if call waiting is enabled.
The call waiting on code and call waiting off code configured on IP phones are used to activate/deactivate the server-side call waiting feature. They may vary on different servers.
Procedure
Call waiting and call waiting tone can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure call waiting.
Parameters: call_waiting.enable call_waiting.tone call_waiting.on_code call_waiting.off_code
Configure call waiting.
Navigate to: http://<phoneIPAddress>/servlet
Configuring Basic Features
Phone User Interface
Details of Configuration Parameters:
?p=features-general&q=load
Configure call waiting.
Parameters Permitted Values Default call_waiting.enable 0 or 1 1
Description:
Enables or disables call waiting feature.
0-Disabled
1-Enabled
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.
Web User Interface:
Features->General Information->Call Waiting
Phone User Interface:
Menu->Features->Call Waiting->Call Waiting call_waiting.tone 0 or 1 1 call_waiting.on_code String within 32 characters Blank
Description:
Configures the call waiting on code to activate the server-side call waiting feature.
The IP phone will send the call waiting on code to the server when you activate call waiting feature on the IP phone.
Example: call_waiting.on_code = *72
Web User Interface:
Features->General Information->Call Waiting On Code
Phone User Interface:
Menu->Features->Call Waiting->On Code call_waiting.off_code String within 32 characters Blank
Description:
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Parameters Permitted Values Default
Configures the call waiting off code to deactivate the server-side call waiting feature. The IP phone will send the call waiting off code to the server when you deactivate call waiting feature on the IP phone.
Example: call_waiting.off_code = *73
Web User Interface:
Features->General Information->Call Waiting Off Code
Phone User Interface:
Menu->Features->Call Waiting->Off Code
To configure call waiting via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Call Waiting.
3. (Optional.) Enter the call waiting on code in the Call Waiting On Code field.
4. (Optional.) Enter the call waiting off code in the Call Waiting Off Code field.
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5. Click Confirm to accept the change.
To configure the call waiting tone via web user interface:
1. Click on Features->Audio.
Configuring Basic Features
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
Tone field.
4. (Optional.) Enter the call waiting on code in the On Code field.
5. (Optional.) Enter the call waiting off code in the Off Code field.
6. Press the Save soft key to accept the change.
Auto redial allows IP phones to redial a busy number after the first attempt. Both the number of attempts and waiting time between redials are configurable.
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.
Parameters: auto_redial.enable auto_redial.interval auto_redial.times
Configure auto redial feature.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure auto redial.
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Details of Configuration Parameters:
Parameters Permitted Values Default auto_redial.enable 0 or 1 0
Description:
Enables or disables the IP phone to automatically redial the dialed number when the callee is temporarily unavailable.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the IP phone will dial the previous dialed out number automatically when the dialed number is temporarily unavailable.
Web User Interface:
Features->General Information->Auto Redial
Phone User Interface:
Menu->Features->Others->Auto Redial->Auto Redial auto_redial.interval Integer from 1 to 300 10
Description:
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.
Web User Interface:
Features->General Information->Auto Redial Interval (1~300s)
Phone User Interface:
Menu->Features->Others->Auto Redial->Redial Interval auto_redial.times Integer from 1 to 300 10
Description:
Configures the auto redial times when the callee is temporarily unavailable.
The IP phone tries to redial the dialed number as many times as configured till the callee answers the call.
Web User Interface:
Features->General Information->Auto Redial Times (1~300)
Phone User Interface:
Menu->Features->Others->Auto Redial->Redial Times
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Configuring Basic Features
To configure auto redial via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Auto Redial.
3. Enter the desired time interval (in seconds) in the Auto Redial Interval (1~300s) field.
The default waiting time is 10s.
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->Others->Auto Redial.
2. Press or , or the Switch soft key to select the desired value from the Auto
Redial field.
3. Enter the desired time in the Redial Interval field.
4. Enter the desired times in the Redial Times field.
5. Press the Save soft key to accept the change.
Auto answer allows IP phones to automatically answer an incoming call. IP phones will not automatically answer the incoming call during a call even if auto answer is enabled.
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.
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Procedure
Auto answer can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure auto answer.
Parameter: account.X.auto_answer
Specify a period of delay time for auto answer.
Parameter: features.auto_answer_delay
Configure auto answer.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-basic&q=load&acc
=0
Specify a period of delay time for auto answer. http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure auto answer.
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.auto_answer 0 or 1 0
Description:
Enables or disables auto answer feature for account X.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the IP phone can automatically answer an incoming call.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Note: The IP phone cannot automatically answer the incoming call during a call even if auto answer is enabled.
Web User Interface:
Account->Basic->Auto Answer
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Configuring Basic Features
Parameters Permitted Values
Phone User Interface:
Menu->Features ->Auto Answer->Line X->Auto Answer features.auto_answer_delay
(X ranges from 1 to 16)
Integer from 1 to 4
Default
1
Description:
Configures the delay time (in seconds) before the IP phone automatically answers an incoming call.
Web User Interface:
Features->General Information->Auto-Answer Delay (1~4s)
Phone User Interface:
None
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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2. Enter the desired time (in seconds) in the Auto-Answer Delay (1~4s) field.
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3. Click Confirm to accept the change.
To configure auto answer via phone user interface:
1. Press Menu->Features->Auto Answer.
2. Select the desired line 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.
Configuring Basic Features
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.
Parameter: features.call_completion_enable
Configure call completion.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure call completion.
Details of the Configuration Parameter:
Parameter Permitted Values Default features.call_completion_enable 0 or 1 0
Description:
Enables or disables call completion feature. If a user places a call and the callee is temporarily unavailable to answer the call, call completion feature allows notifying the user when the callee becomes available to receive a call.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the caller is notified when the callee becomes available to receive a call.
Web User Interface:
Features->General Information->Call Completion
Phone User Interface:
Menu->Features->Others->Call Completion
To configure call completion 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 Call Completion.
3. Click Confirm to accept the change.
To configure call completion via phone user interface:
1. Press Menu->Features->Others->Call Completion.
2. Press or , or the Switch soft key to select the desired value from the Call
Completion field.
3. Press the Save soft key to accept the change.
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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
Configuring Basic Features
User-Agent: Yealink SIP-T46G 28.73.0.10
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 call 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.
Parameters: account.X.anonymous_call account.X.send_anonymous_code account.X.anonymous_call_oncode account.X.anonymous_call_offcode
Configure anonymous call.
Navigate to: http://<phoneIPAddress>/servlet?p
=account-basic&q=load&acc=0
Configure anonymous call.
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.anonymous_call 0 or 1 0
Description:
Enables or disables anonymous call feature for account X.
0-Disabled
1-Enabled
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.
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Parameters Permitted Values
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Basic->Local Anonymous
Phone User Interface:
Menu->Features->Anonymous->Line X ->Local Anonymous account.X.send_anonymous_code 0 or 1
Default
0
Description:
Configures the IP phone to send anonymous on/off code to activate/deactivate the server-side anonymous call feature for account X.
0-Off Code
1-On Code
If it is set to 0 (Off Code), the IP phone will send off code to deactivate the server-side anonymous call feature when you enable or disable local anonymous feature on the phone.
If it is set to 1 (On Code), the IP phone will send on code to activate the server-side anonymous call feature when you enable or disable local anonymous feature on the phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Basic->Send Anonymous Code
Phone User Interface:
Menu->Features->Anonymous->Line X->Send Anonymous Code account.X.anonymous_call_oncode String within 32 characters Blank
Description:
Configures the anonymous call on code to activate the server-side anonymous call feature for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
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Configuring Basic Features
Parameters Permitted Values Default
Example: account.1.anonymous_call_oncode = *72
Note: It works only if the value of the parameter “account.X.send_anonymous_code” is set to 1 (On Code).
Web User Interface:
Account->Basic->Send Anonymous Code->On Code
Phone User Interface:
Menu->Features->Anonymous->Line X ->On Code account.X.anonymous_call_offcode String within 32 characters Blank
Description:
Configures the anonymous call off code to deactivate the server-side anonymous call feature for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.anonymous_call_offcode = *73
Note: It works only if the value of the parameter “account.X.send_anonymous_code” is set to 0 (Off Code).
Web User Interface:
Account->Basic->Send Anonymous Code->Off Code
Phone User Interface:
Menu->Features->Anonymous->Line X ->Off Code
To configure the anonymous call via web user interface:
1. Click on Account.
2. Select the desired account from the pull-down list of Account.
3. Click on Basic.
4. Select the desired value from the pull-down list of Local Anonymous.
5. (Optional.) 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.
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7. (Optional.) Enter the anonymous call off code in the Off Code field.
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8. Click Confirm to accept the change.
To configure the anonymous call via phone user interface:
1. Press Menu->Features->Anonymous.
2. Select the desired line and then press Enter soft key.
3. Press or , or the Switch soft key to select the desired value from the Local
Anonymous field.
4. (Optional.) Press or , or the Switch soft key to select the desired value from the Send Anonymous Code field.
5. (Optional.) Enter the anonymous call on code in the On Code field.
6. (Optional.) Enter the anonymous call off code in the Off Code field.
7. Press the Save soft key to accept the change.
Anonymous call rejection allows IP phones to automatically reject incoming calls from callers whose identity has been deliberately concealed. The anonymous caller’s LCD screen presents “Anonymity Disallowed”. Anonymous call rejection is configurable on a per-line basis.
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. Send Anonymous Rejection Code feature allows IP phones to send anonymous call rejection on/off code to the server.
Configuring Basic Features
Procedure
Anonymous call rejection can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Configure anonymous call rejection.
Parameters: account.X.reject_anonymous_call account.X.send_anonymous_rejection
_code account.X.anonymous_reject_oncode account.X.anonymous_reject_offcode
Web User Interface
Configure anonymous call rejection.
Navigate to: http://<phoneIPAddress>/servlet?p=a ccount-basic&q=load&acc=0
Phone User Interface Configure anonymous call rejection.
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.reject_anonymous_call 0 or 1 0
Description:
Enables or disables anonymous call rejection feature for account X.
0-Disabled
1-Enabled
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Basic->Local Anonymous Rejection
Phone User Interface:
Menu->Features->Anonymous->Line X->Local Anonymous Rejection account.X.send_anonymous_rejection_code 0 or 1 0
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Parameters Permitted Values Default
Description:
Configures the IP phone to send anonymous rejection on/off code to activate/deactivate the server-side anonymous rejection feature for account X.
0-Off Code
1-On Code
If it is set to 0 (Off Code), the IP phone will send reject off code to deactivate the server-side anonymous rejection feature when you enable or disable anonymous rejection feature on the phone.
If it is set to 1 (On Code), the IP phone will send reject on code to activate the server-side anonymous rejection feature when you enable or disable anonymous rejection feature on the phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Basic->Send Anonymous Rejection Code
Phone User Interface:
Menu->Features->Anonymous->Line X->Send Anonymous Rejection Code account.X.anonymous_reject_oncode
String within 32 characters
Blank
Description:
Configures the anonymous call rejection on code to activate the server-side anonymous call rejection feature for account X. The IP phone will send the anonymous call rejection on code to the server when you activate anonymous call rejection feature for account X
on the IP phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.anonymous_reject_oncode = *74
Web User Interface:
Account->Basic->Send Anonymous Rejection Code->On Code
Phone User Interface:
Menu->Features->Anonymous->Line X ->Send Anonymous Rejection Code->On
Code
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Configuring Basic Features
Parameters Permitted Values Default account.X.anonymous_reject_offcode String within 32 characters Blank
Description:
Configures the anonymous call rejection off code to deactivate the server-side anonymous call rejection feature for account X. The IP phone will send the anonymous call rejection off code to the server when you deactivate anonymous call rejection feature for account X
on the IP phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.anonymous_reject_offcode = *75
Web User Interface:
Account->Basic->Send Anonymous Rejection Code->Off Code
Phone User Interface:
Menu->Features->Anonymous->Line X->Send Anonymous Rejection Code->Off
Code
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.) Select the desired value from the pull-down list of Send Anonymous
Rejection Code.
6. (Optional.) Enter the anonymous call rejection on code in the On Code field.
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7. (Optional.) Enter the anonymous call rejection off code in the Off Code field.
8. Click Confirm to accept the change.
To configure anonymous call rejection via phone user interface:
1. Press Menu->Features->Anonymous.
2. Select the desired line and then press Enter soft key.
3. Press or , or the Switch soft key to select the desired value from the Local
Anonymous Rejection field.
4. (Optional.) Press or to select the desired value from the Send Anonymous
Rejection Code field.
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. 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 can be configured on a phone or per-line basis depending on the DND mode. Two DND modes:
Phone (default): DND feature is effective for all accounts on the IP phone.
Custom: DND feature can be configured for each or all accounts.
A user can activate or deactivate DND using the DND soft key or DND key. The DND configurations on IP phones may be overridden by the server settings. The server-side
DND feature disables the local DND and call forward settings. If the server-side DND feature is enabled on any of the IP phone’s registrations, the other registrations are not affected. For more information on call forward, refer to
on page
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.
Configuring Basic Features
Return Message When DND
This feature defines the return code and the reason of the SIP response message for the rejected incoming call when DND is enabled on IP phones. The caller’s LCD screen displays the received return code.
Procedure
DND can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Configure DND in the custom mode.
Parameters: account.X.dnd.enable account.X.dnd.on_code account.X.dnd.off_code
Assign a DND key.
Parameters: linekey.X.type/ programablekey.X.type
Configure the DND mode.
Parameter: features.dnd_mode
Configure DND in the phone mode.
Parameters: features.dnd.enable features.dnd.on_code features.dnd.off_code
Specify return code and reason of the SIP response message.
Parameter: features.dnd_refuse_code
Assign a DND key.
Navigate to: http://<phoneIPAddress>/servlet?
p=dsskey&model=1&q=load&line page=1
Configure DND.
Navigate to: http://<phoneIPAddress>/servlet?
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Phone User Interface p=features-forward&q=load
Specify return code and reason of the SIP response message.
Navigate to: http://<phoneIPAddress>/servlet?
p=features-general&q=load
Assign a DND key.
Configure DND.
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.dnd.enable 0 or 1 0
Description:
Enables or disables DND feature for account X when the DND mode is configured as
Custom.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the IP phone will reject incoming calls on account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Features->Forward& DND->DND ->DND Status
Phone User Interface:
Menu->Features->DND->Account->DND Enable. account.X.dnd.on_code String within 32 characters Blank
Description:
Configures the DND on code to activate the server-side DND feature for account X when the DND mode is configured as Custom. The IP phone will send the DND on code to the server when you activate DND feature for account X on the IP phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.dnd.on_code = *73
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Configuring Basic Features
Parameters Permitted Values
Web User Interface:
Features->Forward& DND->DND->DND On Code
Phone User Interface:
Menu->Features->DND->Account->On Code account.X.dnd.off_code String within 32 characters
Default
Blank
Description:
Configures the DND off code to deactivate the server-side DND feature for account X when the DND mode is configured as Custom. The IP phone will send the DND off code to the server when you deactivate DND feature for account X on the IP phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.dnd.off_code = *74
Web User Interface:
Features->Forward& DND->DND ->DND Off Code
Phone User Interface:
Menu->Features->DND->Account->Off Code features.dnd_mode 0 or 1 0
Description:
Configures the DND mode for the IP phone.
0-Phone
1-Custom
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.
Web User Interface:
Features->Forward& DND->DND->Mode
Phone User Interface:
None features.dnd.enable 0 or 1 0
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Parameters Permitted Values Default
Description:
Enables or disables DND feature when the DND mode is configured as Phone.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the IP phone will reject incoming calls on all accounts.
Web User Interface:
Features->Forward& DND->DND->DND Status
Phone User Interface:
Menu->Features->DND->DND Enable features.dnd.on_code String within 32 characters Blank
Description:
Configures the DND on code to activate the server-side DND feature when the DND mode is configured as Phone. The IP phone will send the DND on code to the server when you activate DND feature on the IP phone.
Example: features.dnd.on_code = *71
Web User Interface:
Features->Forward& DND->DND->On Code
Phone User Interface:
Menu->Features->DND->On Code features.dnd.off_code String within 32 characters Blank
Description:
Configures the DND off code to deactivate the server-side DND feature when the
DND mode is configured as Phone. The IP phone will send the DND off code to the server when you deactivate DND feature on the IP phone.
Example: features.dnd.off_code = *72
Web User Interface:
Features->Forward& DND->DND->DND Off Code
Phone User Interface:
Menu->Features->DND->Off Code features.dnd_refuse_code 404, 480 or 486 480
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Configuring Basic Features
Parameters Permitted Values Default
Description:
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.
404-No Found
480-Temporarily Unavailable
486-Busy Here
If it is set to 486 (Busy Here), the caller’s LCD screen will display the reason “Busy
Here” when the callee enables DND.
Web User Interface:
Features->General Information->Return Code When DND
Phone User Interface:
None
DND Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameter
Permitted
Values
Default linekey.X.type
/programablekey.X.type
5
Refer to the following content
Description:
Configures a DSS key as a DND key on the IP phone.
The digit 5 stands for the key type DND.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Example: linekey.2.type = 5
Default:
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Parameter
Permitted
Values
Default
For line keys:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
For programable keys:
For SIP-T48G/T46G IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 (
Directory
).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61 (
Directory
).
When X=7, the default value is 51 (
Switch Account Up
).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=12, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
When X=14, the default value is 2 (
Forward
).
For SIP-T42G/T41P IP phones:
When X=1, the default value is 28 ( History ).
When X=2, the default value is 61 ( Directory ).
When X=3, the default value is 5 ( DND ).
When X=4, the default value is 30 ( Menu ).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61 (
Directory
).
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Configuring Basic Features
Parameter
Permitted
Values
When X=7, the default value is 51 ( Switch Account Up ).
When X=8, the default value is 52 ( Switch Account Down ).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 ( NA ).
When X=13, the default value is 0 (
NA
).
Web User Interface:
DSSKey->Line Key X
/ Programable Key
->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type
Default
To configure a DND key via web user interface:
1. Click on DSSKey->Line Key X.
2. In the desired DSS key field, select DND from the pull-down list of Type.
3. Click Confirm to accept the change.
To configure the DND feature via web user interface:
1. Click on Features->Forward & DND.
2. In the DND block, mark the desired radio box in the Mode field. a) If you mark the Phone radio box:
1) Mark the desired radio box in the DND Status field.
2) (Optional.) Enter the DND on code in the DND On Code field.
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3) (Optional.) Enter the DND off code in the DND Off Code field. b) If you mark the Custom radio box:
1) Select the desired account from the pull-down list of Account.
2) Mark the desired value in the DND Status field.
3) (Optional.) Enter the DND on code in the DND On Code field.
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Configuring Basic Features
4) (Optional.) Enter the DND off code in the DND Off Code field.
3. Click Confirm to accept the change.
To specify the return code via web user interface:
1. Click on Features->General Information.
2. Select the desired type from the pull-down list of Return Code When DND.
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3. Click Confirm to accept the change.
To configure a DND key via phone user interface:
1. Press Menu->Features->DSS Keys.
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select DND from the Key Type field.
5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Press the Save soft key to accept the change.
To configure DND in the phone mode via phone user interface:
1. Press the DND soft key or the DND key when the IP phone is idle.
To configure DND in the custom mode for a specific account via phone user interface:
1. Press the DND soft key or the DND key when the IP phone is idle.
The LCD screen displays a list of the accounts registered on the IP phone.
2. Press or to select the desired account and press the Enter soft key.
3. Press or to select On to activate DND.
4. Press the Save soft key to accept the change.
To configure DND in the custom mode for all accounts via phone user interface:
1. Press the DND soft key or the DND key when the IP phone is idle.
The LCD screen displays a list of the accounts registered on the IP phone.
2. Press the All On soft key to activate DND for all accounts.
3. Press the Save soft key to accept the change.
172
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 the busy tone delay feature.
Parameter: features.busy_tone_delay
Configure the busy tone delay feature.
Configuring Basic Features
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default features.busy_tone_delay 0, 3 or 5 0
Description:
Configures the duration time (in seconds) for the busy tone.
When one party releases the call, a busy tone is audible to the other party indicating that the call connection breaks.
0-0s
3-3s
5-5s
If it is set to 3 (3s), a busy tone is audible for 3 seconds on the IP phone.
Web User Interface:
Features->General Information->Busy Tone Delay (Seconds)
Phone User Interface:
None
To configure busy tone delay via web user interface:
1． Click on Features->General Information.
2． Select the desired value from the pull-down list of Busy Tone Delay (Seconds).
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3． Click Confirm to accept the change.
174
Return code when refuse defines the return code and reason of the SIP response message for call rejection. The caller’s LCD screen displays the reason according to the return code received. Available return codes and reasons are:
404 (Not Found)
480 (Temporarily Unavailable)
486 (Busy Here)
Procedure
Return code for call rejection can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the return code when refusing a call.
Parameter: features.normal_refuse_code
Configure the return code when refusing a call.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default features.normal_refuse_code 404, 480 or 486 486
Description:
Configures a return code and reason of SIP response messages when the IP phone rejects an incoming call. A specific reason is displayed on the caller’s phone LCD screen.
404-No Found
480-Temporarily Unavailable
486-Busy Here
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.
Web User Interface:
Configuring Basic Features
Parameter Permitted Values Default
Features->General Information->Return Code When Refuse
Phone User Interface:
None
To specify the return code when refusing a call via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Return Code When Refuse.
3. Click Confirm to accept the change.
Early media refers to media (e.g., audio and video) played to the caller before a SIP call is actually established. Current implementation supports early media through the
183 message. When the caller receives a 183 message with SDP before the call is established, a media channel is established. This channel is used to provide the early media stream for the caller.
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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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.
Parameter: phone_setting.is_deal180
Configure 180 ring workaround.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default phone_setting.is_deal180 0 or 1 1
Description:
Enables or disables the IP phone to deal with the 180 SIP message received after the
183 SIP message.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the IP phone will resume and play the local ringback tone upon a subsequent 180 message received.
Web User Interface:
Features->General Information->180 Ring Workaround
Phone User Interface:
None
To configure 180 ring workaround 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 180 Ring Workaround.
3. Click Confirm to accept the change.
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.
Parameter: sip.use_out_bound_in_dialog
Specify whether to use outbound proxy in a dialog.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
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Details of the Configuration Parameter:
Parameter Permitted Values Default sip.use_out_bound_in_dialog 0 or 1 1
Description:
Enables or disables the IP phone to keep sending SIP requests to the outbound proxy server in a dialog.
0-Disabled
1-Enabled
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 in a dialog.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Features->General Information->Use Outbound Proxy In Dialog
Phone User Interface:
None
To specify whether to use outbound proxy server in a dialog via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Use Outbound Proxy In Dialog.
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3. Click Confirm to accept the change.
Configuring Basic Features
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 retransmitting time of any SIP request message. The re-transmitting and doubling of T1 will continue until the retransmitting time reaches the T2 value. Timer T4 represents the time the network will take to clear messages between the SIP client and server. 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.
Parameters: account.X.advanced.timer_t1 account.X.advanced.timer_t2 account.X.advanced.timer_t4
Configure SIP session timer.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.advanced.timer_t1 Float from 0.5 to10 0.5
Description:
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->SIP Session Timer T1 (0.5~10s)
Phone User Interface:
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Parameters
None account.X.advanced.timer_t2
Permitted Values Default
Float from 2 to 40 4
Description:
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->SIP Session Timer T2 (2~40s)
Phone User Interface:
None account.X.advanced.timer_t4 Float from 2.5 to 60 5
Description:
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->SIP Session Timer T4 (2.5~60s)
Phone User Interface:
None
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.
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Configuring Basic Features
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.
Call hold provides a service of placing an active call on hold. When a call is placed on hold, the IP phones send an INVITE request with HOLD SDP to request remote parties to stop sending media and to inform them that they are being held. 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 warning tone at regular intervals when there is a call on hold. The warning tone is played through the speakerphone.
IP phones also support Music on Hold (MoH) feature. 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 a 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
Call hold can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Configure the call hold tone and call hold tone delay.
Parameters: features.play_hold_tone.enable
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Local
<MAC>.cfg
Web User Interface features.play_hold_tone.delay
Specify whether RFC 2543
(c=0.0.0.0) outgoing hold signaling is used.
Parameters: sip.rfc2543_hold
Configure MoH on a per-line basis.
Parameter: account.X.music_server_uri
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
Configure MoH on a per-line basis.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
Details of Configuration Parameters:
Parameters Permitted Values Default features.play_hold_tone.enable 0 or 1
Description:
Enables or disables the IP phone to play a tone when there is a call on hold.
0-Disabled
1-Enabled
Web User Interface:
Features->General Information->Play Hold Tone
Phone User Interface:
None
1
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Configuring Basic Features
Parameters Permitted Values Default features.play_hold_tone.delay Integer from 3 to 3600 30
Description:
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 value of the parameter “features.play_hold_tone.enable” is set to 1 (Enabled).
Web User Interface:
Features->General Information->Play Hold Tone Delay
Phone User Interface:
None sip.rfc2543_hold 0 or 1 0
Description:
Enables or disables the IP phone to use RFC 2543 (c=0.0.0.0) outgoing hold signali ng.
0-Disabled
1-Enabled
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.
Web User Interface:
Features->General Information->RFC 2543 Hold
Phone User Interface:
None account.X.music_server_uri SIP URI within 256 characters Blank
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Parameters Permitted Values Default
Description:
Configures the address of the Music On Hold server for account X. 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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.music_server_uri = sip:[email protected]
Note: The DNS query in this parameter only supports A query.
Web User Interface:
Account->Advanced->Music Server URI
Phone User Interface:
None
To configure call hold method via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of RFC 2543 Hold.
184
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.
Configuring Basic Features
2. Select the desired value from the pull-down list of Play Hold Tone.
3. Enter the desired time in the Play Hold Tone Delay field.
4. Click Confirm to accept the change.
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.
4. Enter the SIP URI (e.g., sip:[email protected]) in the Music Server URI field.
5. Click Confirm to accept the change.
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186
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 request at or before the negotiated session expiration.
Procedure
Session timer can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure session timer.
Parameters: account.X.session_timer.enable account.X.session_timer.expires account.X.session_timer.refresher
Configure session timer.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.session_timer.enable 0 or 1 0
Description:
Enables or disables the session timer for account X.
0-Disabled
1-Enabled
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Configuring Basic Features
Parameters
Account->Advanced->Session Timer
Phone User Interface:
None account.X.session_timer.expires
Permitted Values Default
Integer from 30 to 7200 1800
Description:
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 refresh the session during a call before
1800 seconds.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.session_timer.expires = 1800
Web User Interface:
Account->Advanced->Session Expires(30~7200s)
Phone User Interface:
None account.X.session_timer.refresher 0 or 1 0
Description:
Configures the session timer refresher for account X.
0-UAC
1-UAS
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->Session Refresher
Phone User Interface:
None
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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. 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.
188
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 calls immediately.
Busy Forward -- Forward the incoming call when the IP phone or the specified account is busy.
No Answer Forward -- Forward the incoming call after a period of ring time.
Call forward can be configured on a phone or per-line basis depending on the call forward mode. The following describes the call forward modes:
Phone (default): Call forward feature is effective for all accounts on the IP phone.
Configuring Basic Features
Custom: Call forward feature can be configured for each or all accounts.
The server-side call forward settings disable the local call forward settings. If the server-side call forward feature is enabled on any of the IP phone’s registrations, the other registrations are not affected. DND activated on the IP phone disables the local no answer forward settings.
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.
IP phones support the redirected call information sent by the SIP server with Diversion header, per draft-levy-sip-diversion-08, or History-info header, per RFC 4244. The
Diversion/History-info header is used to inform the IP phone of a call’s history. For example, when a phone has been set to enable call forward, the Diversion/History-info header allows the receiving phone to indicate who the call was from, and from which phone number it was forwarded.
Forward International
Forward international allows users to forward an incoming call to an international telephone number. This feature is enabled by default.
Procedure
Call forward can be configured using the configuration files or locally.
Configuration File <MAC>.cfg
Configure call forward in custom mode.
Parameters: account.X.always_fwd.enable account.X.always_fwd.target account.X.always_fwd.on_code account.X.always_fwd.off_code account.X.busy_fwd.enable account.X.busy_fwd.target account.X.busy_fwd.on_code account.X.busy_fwd.off_code account.X.timeout_fwd.enable account.X.timeout_fwd.target account.X.timeout_fwd.timeout account.X.timeout_fwd.on_code account.X.timeout_fwd.off_code
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Local
<y0000000000xx>.cf
g
Web User Interface
Phone User Interface
Configure the call forward mode.
Parameter: features.fwd_mode
Configure call forward in phone mode.
Parameters: 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 features.fwd_diversion_enable
Configure forward international.
Parameter: forward.international.enable
Configure call forward.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-forward&q=load
Configure forward international.
Navigate to: http://<phoneIPAddress>/ servlet?p=features-general&q=l oad
Configure call forward.
Configure forward international.
190
Configuring Basic Features
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.always_fwd.enable 0 or 1 0
Description:
Enables or disables always forward feature for account X when the call forward mode is configured as Custom
.
0-Disabled
1-Enabled
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Features->Forward& DND->Forward->Always Forward->On/Off
Phone User Interface:
Menu->Features->Call Forward->Account ->Always Forward ->Always Forward account.X.always_fwd.target String within 32 characters Blank
Description:
Configures the destination number of the always forward for account X
when the call forward mode is configured as Custom .
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.always_fwd.target = 3601
Web User Interface:
Features->Forward& DND->Forward->Always Forward->Target
Phone User Interface:
Menu->Features->Call Forward->Account ->Always Forward ->Forward To account.X.always_fwd.on_code String within 32 characters Blank
Description:
Configures the always forward on code to activate the server-side always forward
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Parameters Permitted Values Default feature for account X when the call forward mode is configured as Custom
. The IP phone will send the always forward on code and the pre-configured destination number to the server when you activate always forward feature for account X on the
IP phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.always_fwd.on_code = *72
Web User Interface:
Features->Forward& DND->Forward->Always Forward->On Code
Phone User Interface:
Menu->Features->Call Forward->Account ->Always Forward ->On Code account.X.always_fwd.off_code String within 32 characters Blank
Description:
Configures the always forward off code to deactivate the server-side always forward feature for account X
when the call forward mode is configured as Custom
. The
IP phone will send the always forward off code to the server when you deactivate always forward feature for account X on the IP phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.busy_fwd.off_code = *73
Web User Interface:
Features->Forward& DND->Forward->Always Forward ->Off Code
Phone User Interface:
Menu->Features->Call Forward->Account ->Always Forward ->Off Code account.X.busy_fwd.enable
0 or 1 0
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Configuring Basic Features
Parameters Permitted Values Default
Description:
Enables or disables busy forward feature for account X
when the call forward mode is configured as Custom
.
0-Disabled
1-Enabled
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Features->Forward& DND->Forward->Busy Forward->On/Off
Phone User Interface:
Menu->Features->Call Forward->Account ->Busy Forward->Busy Forward account.X.busy_fwd.target String within 32 characters Blank
Description:
Configures the destination number of the busy forward for account X
when the call forward mode is configured as Custom.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.busy_fwd.target = 3602
Web User Interface:
Features->Forward& DND->Forward->Busy Forward->Target
Phone User Interface:
Menu->Features->Call Forward->Account ->Busy Forward ->Forward To account.X.busy_fwd.on_code
String within 32 characters Blank
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Parameters Permitted Values Default
Description:
Configures the busy forward on code to activate the server-side busy forward feature for account X
when the call forward mode is configured as Custom.
The IP phone will send the busy forward on code and the pre-configured destination number to the server when you activate busy forward feature for account X on the IP phone.
Example: account.1.busy_fwd.on_code = *74
Web User Interface:
Features->Forward& DND->Forward->No Answer Forward ->On Code
Phone User Interface:
Menu->Features->Call Forward->Account ->Busy Forward ->On Code account.X.busy_fwd.off_code String within 32 characters Blank
Description:
Configures the busy forward off code to deactivate the server-side busy forward feature for account X
when the call forward mode is configured as Custom.
The IP phone will send the busy forward off code to the server when you deactivate busy forward feature for account X on the IP phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.busy_fwd.off_code = *75
Web User Interface:
Features->Forward& DND->Forward ->No Answer Forward ->Off Code
Phone User Interface:
Menu->Features->Call Forward->Account ->Busy Forward ->Off Code account.X.timeout_fwd.enable 0 or 1 0
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Configuring Basic Features
Parameters Permitted Values Default
Description:
Enables or disables no answer forward feature for account X
when the call forward mode is configured as Custom.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), incoming calls to the account X are forwarded to the destination number after a period of ring time.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Features->Forward& DND->Forward->No Answer Forward->On/Off
Phone User Interface:
Menu->Features->Call Forward->Account ->No Answer Forward->No Answer
Forward account.X.timeout_fwd.target String within 32 characters Blank
Description:
Configures the destination number of the no answer forward for account X
when the call forward mode is configured as Custom.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.timeout_fwd.target = 3603
Web User Interface:
Features->Forward& DND->Forward->No Answer Forward->Target
Phone User Interface:
Menu->Features->Call Forward->Account ->No Answer Forward ->Forward To account.X.timeout_fwd.timeout Integer from 0 to 20 2
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Parameters Permitted Values Default
Description:
Configures ring times (N) to wait before forwarding incoming calls for account X when the call forward mode is configured as Custom.
Incoming calls will be forwarded when not answered after N*6 seconds.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Features->Forward& DND->Forward->No Answer Forward->After Ring Time
Phone User Interface:
Menu->Features->Call Forward->Account ->No Answer Forward ->After Ring Time account.X.timeout_fwd.on_code String within 32 characters Blank
Description:
Configures the no answer forward on code to activate the server-side no answer forward feature for account X
when the call forward mode is configured as Custom.
The
IP phone will send the no answer forward on code and the pre-configured destination number to the server when you activate no answer forward feature for account X on the IP phone.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.timeout_fwd.on_code = *76
Web User Interface:
Features->Forward& DND->Forward->No Answer Forward ->On Code
Phone User Interface:
Menu->Features->Call Forward->Account ->No Answer Forward ->On Code account.X.timeout_fwd.off_code String within 32 characters Blank
Description:
Configures the no answer forward off code to deactivate the server-side no answer forward feature for account X
when the call forward mode is configured as Custom.
The
IP phone will send the no answer forward off code to the server when you deactivate no answer forward feature for account X on the IP phone.
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Parameters Permitted Values Default
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.timeout_fwd.off_code = *77
Web User Interface:
Features->Forward& DND->Forward->No Answer Forward ->Off Code
Phone User Interface:
Menu->Features->Call Forward->Account ->No Answer Forward ->Off Code features.fwd_mode 0 or 1 0
Description:
Configures the call forward mode for the IP phone.
0-Phone
1-Custom
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.
Web User Interface:
Features->Forward&DND->Forward->Mode
Phone User Interface:
None forward.always.enable 0 or 1 0
Description:
Enables or disables always forward feature.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), incoming calls are forwarded to the destination number immediately.
Web User Interface:
Features->Forward &DND->Forward->Always Forward->On/Off
Phone User Interface:
Menu->Features->Call Forward->Always Forward->Always Forward forward.always.target String within 32 characters Blank
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Parameters Permitted Values Default
Description:
Configures the destination number the IP phone forwards all incoming calls to.
Web User Interface:
Features->Forward &DND->Always Forward->Target
Phone User Interface:
Menu->Features->Call Forward->Always Forward->Forward To forward.always.on_code String within 32 characters Blank
Description:
Configures the always forward on code to activate the server-side always forward feature. The IP phone will send the always forward on code and the pre-configured destination number to the server when you activate always forward feature on the IP phone.
Example: forward.always.on_code = *72
Web User Interface:
Features->Forward &DND->Forward->Always Forward->On Code
Phone User Interface:
Menu->Features->Call Forward->Always Forward->On Code forward.always.off_code String within 32 characters Blank
Description:
Configures the always forward off code to deactivate the server-side always forward feature. The IP phone will send the always forward off code to the server when you deactivate always forward feature on the IP phone.
Example: forward.always.off_code = *73
Web User Interface:
Features->Forward &DND->Forward->Always Forward->Off Code
Phone User Interface:
Menu->Features->Call Forward->Always Forward->Off Code forward.busy.enable 0 or 1 0
Description:
Enables or disables busy forward feature.
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Parameters Permitted Values Default
0-Disabled
1-Enabled
If it is set to 1 (Enabled), incoming calls are forwarded to the destination number when the callee is busy.
Web User Interface:
Features->Forward &DND->Forward->Busy Forward->On/Off
Phone User Interface:
Menu->Features->Call Forward->Busy Forward->Busy Forward forward.busy.target String within 32 characters Blank
Description:
Configures the destination number the IP phone forwards incoming calls to when busy.
Example: forward.busy.target = 3602
Web User Interface:
Features->Forward &DND->Forward->Busy Forward->Target
Phone User Interface:
Menu->Features->Call Forward->Busy Forward->Forward To forward.busy.on_code String within 32 characters Blank
Description:
Configures the busy forward on code to activate the server-side busy forward feature. The IP phone will send the busy forward on code and the pre-configured destination number to the server when you activate busy forward feature on the IP phone.
Example: forward.busy.on_code = *74
Web User Interface:
Features->Forward &DND->Forward->Busy Forward->On Code
Phone User Interface:
Menu->Features->Call Forward->Busy Forward->On Code forward.busy.off_code String within 32 characters Blank
Description:
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Parameters Permitted Values Default
Configures the busy forward off code to deactivate the server-side busy forward feature. The IP phone will send the busy forward off code to the server when you deactivate busy forward feature on the IP phone.
Example: forward.busy.off_code = *75
Web User Interface:
Features->Forward &DND->Forward->Busy Forward->Off Code
Phone User Interface:
Menu->Features->Call Forward->Busy Forward->Off Code forward.no_answer.enable 0 or 1 0
Description:
Enables or disables no answer forward feature.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), incoming calls are forwarded to the destination number after a period of ring time.
Web User Interface:
Features->Forward &DND->Forward->No Answer Forward->On/Off
Phone User Interface:
Menu->Features->Call Forward->No Answer Forward->No Answer Forward forward.no_answer.target String within 32 characters Blank
Description:
Configures the destination number the IP phone forwards incoming calls to after a period of ring time.
Example: forward.no_answer.target = 3603
Web User Interface:
Features->Forward &DND->Forward->No Answer Forward->Target
Phone User Interface:
Menu->Features->Call Forward->No Answer Forward->Forward To forward.no_answer.timeout Integer from 0 to 20 2
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Configuring Basic Features
Parameters Permitted Values Default
Description:
Configures ring times (N) to wait before forwarding incoming calls.
Incoming calls will be forwarded when not answered after N*6 seconds.
Web User Interface:
Features->Forward &DND->Forward->No Answer Forward->After Ring Time
(0~120s)
Phone User Interface:
Menu->Features->Call Forward->No Answer Forward->After Ring Time forward.no_answer.on_code String within 32 characters Blank
Description:
Configures the no answer forward on code to activate the server-side no answer forward feature. The IP phone will send the no answer forward on code and the pre-configured destination number to the server when you activate no answer forward feature on the IP phone.
Example: forward.no_answer.on_code = *76
Web User Interface:
Features->Forward &DND->Forward->No Answer Forward->On Code
Phone User Interface:
Menu->Features->Call Forward->No Answer Forward->On Code forward.no_answer.off_code String within 32 characters Blank
Description:
Configures the no answer forward off code to deactivate the server-side no answer forward feature. The IP phone will send the no answer forward off code to the server when you deactivate no answer forward feature on the IP phone.
Example: forward.no_answer.off_code = *77
Web User Interface:
Features->Forward &DND->Forward->No Answer Forward->Off Code
Phone User Interface:
Menu->Features->Call Forward->No Answer Forward->Off Code features.fwd_diversion_enable 0 or 1 1
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Parameters Permitted Values Default
Description:
Enables or disables the IP phone to present the diversion information when an incoming call is forwarded to your IP phone.
0-Disabled
1-Enabled
Web User Interface:
Features->General Information->Diversion/History-Info
Phone User Interface:
None forward.international.enable 0 or 1 1
Description:
Enables or disables the IP phone to forward incoming calls to international numbers
(the prefix is 00).
0-Disabled
1-Enabled
Web User Interface:
Features->General Information->Fwd International
Phone User Interface:
None
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 Forward/Busy Forward/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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Configuring Basic Features
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 Forward/Busy Forward/No
Answer Forward field.
3) Enter the destination number you want to forward in the Target field.
4) (Optional.) 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 no answer forward).
3. Click Confirm to accept the change.
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To configure the forward international feature via web user interface:
1. Click on Features->General Information.
2. Select the desired value from the pull-down list of Fwd International.
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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 Forward 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 Forward 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.
Configuring Basic Features c) If you select No Answer Forward:
1) Press or , or the Switch soft key to select the desired value from the
No Answer Forward 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 Forward 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 Forward 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 Forward 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.
You can also configure the busy forward for all accounts. After the busy forward was configured for a specific account, do the following:
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1) Press or to highlight the Busy Forward 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 Forward 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 Forward 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.
To configure forward international via phone user interface:
1. Press Menu->Advanced (default password: admin) ->FWD International.
2. Press or , or the Switch soft key to select the desired type from the FWD
International field.
3. Press the Save soft key to accept the change.
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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
Configuring Basic Features 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 attended transfer on hook features allow the IP phone to complete the transfer through on-hook.
When a user performs a semi-attended transfer, semi-attended transfer determines whether to display the prompt “1 New Missed Call(s)” ("n" indicates the number of the missed calls) on the destination party’s LCD screen.
Procedure
Call transfer can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.c
fg
Web User Interface
Specify whether to complete the transfer through on-hook.
Parameters: transfer.blind_tran_on_hook_enable transfer.on_hook_trans_enable
Configure the semi-attended transfer feature.
Parameter: transfer.semi_attend_tran_enable
Specify whether to complete the transfer through on-hook.
Configure the semi-attended transfer feature.
Navigate to: http://<phoneIPAddress>/servlet?p
=features-transfer&q=load
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Details of Configuration Parameters:
Parameters Permitted Values Default transfer.blind_tran_on_hook_enable 0 or 1 1
Description:
Enables or disables the IP phone to complete the blind transfer through on-hook besides pressing the Transfer/Tran soft key or TRAN/TRANSFER key.
0-Disabled
1-Enabled
Web User Interface:
Features->Transfer ->Blind Transfer On Hook
Phone User Interface:
None transfer.on_hook_trans_enable 0 or 1 1
Description:
Enables or disables the IP phone to complete the semi-attended/attended transfer through on-hook besides pressing the Transfer/Tran soft key or TRAN/TRANSFER key.
0-Disabled
1-Enabled
Web User Interface:
Features->Transfer ->Attended Transfer On Hook transfer.semi_attend_tran_enable 0 or 1 1
Description:
Enables or disables the transferee party’s phone to prompt a missed call on the LCD screen before displaying the caller ID when performing a semi-attended transfer.
0-Disabled
1-Enabled
Web User Interface:
Features->Transfer ->Semi-Attended Transfer
Phone User Interface:
None
To configure call transfer via web user interface:
1. Click on Features->Transfer.
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Configuring Basic Features
2. Select the desired values from the pull-down lists of Semi-Attend Transfer, Blind
Transfer On Hook and Semi Attend Transfer On Hook.
3. Click Confirm to accept the change.
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.
Parameters: account.X.conf_type account.X.conf_uri
Configure network conference.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.conf_type 0 or 2 0
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Parameters Permitted Values Default
Description:
Configures the network conference type for account X.
0-Local Conference
2-Network Conference
If it is set to 0 (Local Conference), conferences are set up on the IP phone locally.
If it is set to 2 (Network Conference), conferences are set up by the server.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->Conference Type
Phone User Interface:
None account.X.conf_uri SIP URI within 511 characters Blank
Description:
Configures the network conference URI for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.conf_uri = [email protected]
Note: It works only if the value of the parameter “account.X.conf_type” is set to 2
(Network Conference).
Web User Interface:
Account->Advanced->Conference URI
Phone User Interface:
None
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.
210
5. Enter the conference URI in the Conference URI field.
Configuring Basic Features
6. Click Confirm to accept the change.
Feature key synchronization provides the capability to synchronize the status of the following features between the IP phone and the server:
Do Not Disturb (DND)
Call Forwarding Always (CFA)
Call Forwarding Busy (CFB)
Call Forwarding No Answer (CFNA)
Note
Feature key synchronization is applicable to IP phones running firmware version 73 or later in the neutral version.
Procedure
Feature key synchronization can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure feature key synchronization.
Parameters: bw.feature_key_sync
Configure feature key synchronization.
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Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Details of Configuration Parameter:
Parameters Permitted Values Default bw.feature_key_sync 0 or 1
Description:
Enables or disables feature key synchronization.
0-Disabled
1-Enabled
Web User Interface:
Features->General Information ->Feature Key Synchronization
Phone User Interface:
None
To configure feature key synchronization via web user interface:
1. Click on Features->General Information.
2. Select Enabled from the pull-down list of Feature Key Synchronization.
0
212
3. Click Confirm to accept the change.
Configuring Basic Features
For local conference, all parties drop the call when the conference initiator drops the conference call. For local conference, 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 feature can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure transfer on conference hang up.
Parameter: transfer.tran_others_after_conf_e
nable
Configure transfer on conference hang up.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-transfer&q=load
Details of the Configuration Parameter:
Parameter & Description Permitted Values Default transfer.tran_others_after_conf_enable 0 or 1
Description:
Enables or disables the IP phone to transfer the local conference call to the two parties after the conference initiator drops the local conference call.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the other two parties remain connected when the conference initiator drops the conference call.
Note: It is only applicable to the local conference.
Web User Interface:
Features->Transfer ->Transfer on Conference Hang up
Phone User Interface:
None
0
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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.
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 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.
Parameter: account.X.direct_pickup_code
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Local
Configuring Basic Features
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Configure directed call pickup features on a phone basis.
Parameters: features.pickup.direct_pickup_
enable features.pickup.direct_pickup_c
ode
Assign a directed call pickup key.
Parameters: linekey.X.type/ programablekey.X.type linekey.X.line/ programablekey.X.line linekey.X.value/ programablekey.X.value
Assign a directed call pickup key.
Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&model=1&q=loa d&linepage=1
Configure the directed call pickup feature on a phone basis.
Navigate to: http://<phoneIPAddress>/servl et?p=features-callpickup&q=lo ad
Configure the 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.
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Details of Configuration Parameters:
Parameters Permitted Values Default account.X.direct_pickup_code
String within 32 characters Blank
Description :
Configures the directed call pickup code for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.direct_pickup_code = *97
Note: The directed call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
Web User Interface:
Account->Advanced->Directed Call Pickup Code
Phone User Interface:
None features.pickup.direct_pickup_enable 0 or 1 0
Description:
Enables or disables the IP phone to display the DPickup soft key when the IP phone is in the pre-dialing screen.
0-Disabled
1-Enabled
Web User Interface:
Features->Call Pickup->Directed Call Pickup
Phone User Interface:
None features.pickup.direct_pickup_code String within 32 characters Blank
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Configuring Basic Features
Parameters Permitted Values Default
Description:
Configures the directed call pickup code on a phone basis.
Example: features.pickup.direct_pickup_code = *97
Note: The directed call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
Web User Interface:
Features->Call Pickup->Directed Call Pickup Code
Phone User Interface:
None
Directed Call Pickup Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters Permitted Values Default linekey.X.type/programablekey.X.type 9
Refer to the following content
Description:
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 line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Example: linekey.2.type = 9
Default:
For line keys:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29
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Parameters Permitted Values Default is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
For programable keys:
For SIP-T48G/T46G IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 ( Directory ).
When X=3, the default value is 5 ( DND ).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61 (
Directory
).
When X=7, the default value is 51 (
Switch Account Up
).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=12, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
When X=14, the default value is 2 (
Forward
).
For
SIP-T42G/T41P
IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 (
Directory
).
When X=3, the default value is 5 ( DND ).
When X=4, the default value is 30 ( Menu ).
When X=5, the default value is 28 ( History ).
When X=6, the default value is 61 ( Directory ).
When X=7, the default value is 51 ( Switch Account Up ).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
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Configuring Basic Features
Parameters Permitted Values
When X=10, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
Web User Interface:
DSSKey->Line Key X/ Programable Key ->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type
Default linekey.X.line/programablekey.X.line Integer from 1 to 16
1-6 correspond to the lines 1-6
Description:
Configures the desired line to apply the directed call pickup key.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
The valid values are:
1 to 16 (for SIP-T48G/T46G)
1 to 12 (for SIP-T42G)
1 to 6 (for SIP-T41P)
1-Line 1
2-Line 2
…
16-Line 16
Web User Interface:
DSSKey->Line Key X/ Programable Key ->Line
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Account ID linekey.X.value/ programablekey.X.value
String within 99 characters
Blank
Description:
Configures the directed call pickup feature code followed by the number of monitored extension.
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Parameters Permitted Values
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Web User Interface:
DSSKey->Line Key X/Programable Key ->Value
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Value
Default
To configure a directed call pickup key via web user interface:
1. Click on DSSKey->Line Key X.
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.
4. Select the desired line from the pull-down list of Line.
220
5. Click Confirm to accept the change.
To configure the 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.
Configuring Basic Features
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.
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.
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 DPickup from the Key Type field.
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5. Press or , or the Switch soft key to select the desired line from the Account
ID field.
6. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
7. Enter the directed call pickup code followed by the specific extension in the Value field.
8. 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. 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 per-line basis.
Procedure
Group call pickup can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Configures the group call pickup code on a per-line basis.
Parameter: account.X.group_pickup_code
Configures the group call pickup features on a phone basis.
Parameters: features.pickup.group_pickup_
enable features.pickup.group_pickup_
code
Assign a group call pickup key.
Parameters: linekey.X.type/ programablekey.X.type linekey.X.line/ programablekey.X.line linekey.X.value/ programablekey.X.value
Assign a group call pickup key.
Navigate to: http://<phoneIPAddress>/servl
Configuring Basic Features
Phone User Interface
Details of Configuration Parameters:
Parameters et?p=dsskey&model=1&q=loa d&linepage=1
Configure the 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.
Permitted Values Default features.pickup.group_pickup_enable
0 or 1 0
Description:
Enables or disables the IP phone to display the GPickup soft key when the IP phone is in the pre-dialing screen.
0-Disabled
1-Enabled
Web User Interface:
Features->Call Pickup->Group Call Pickup
Phone User Interface:
None account.X.group_pickup_code String within 32 characters Blank
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Parameters Permitted Values Default
Description:
Configures the group pickup code for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.group_pickup_code = *98
Note: The group call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
Web User Interface:
Account->Advanced->Group Call Pickup Code
Phone User Interface:
None features.pickup.group_pickup_code String within 32 characters Blank
Description:
Configures the group call pickup code on a phone basis.
Example: features.pickup.group_pickup_code = *98
Note: The group call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
Web User Interface:
Features->Call Pickup->Group Call Pickup Code
Phone User Interface:
None
Group Call Pickup Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters linekey.X.type/ programablekey.X.type
Permitted Values
23
Default
Refer to the following content
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Configuring Basic Features
Parameters Permitted Values Default
Description:
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 line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Example: linekey.2.type = 23
Default:
For line keys:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
For programable keys:
For SIP-T48G/T46G IP phones:
When X=1, the default value is 28 ( History ).
When X=2, the default value is 61 ( Directory ).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 ( History ).
When X=6, the default value is 61 (
Directory
).
When X=7, the default value is 51 (
Switch Account Up
).
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Parameters Permitted Values
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=12, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
When X=14, the default value is 2 (
Forward
).
For
SIP-T42G/T41P
IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 (
Directory
).
When X=3, the default value is 5 ( DND ).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 ( History ).
When X=6, the default value is 61 ( Directory ).
When X=7, the default value is 51 ( Switch Account Up ).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
Web User Interface:
DSSKey->Line Key X/Programable Key->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type linekey.X.line/programablekey.X.line
Default
Integer from 1 to 6
1-6 correspond to the lines 1-6.
Description:
Configures the desired line to apply the group call pickup key.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
The valid values are:
1 to 16 (for SIP-T48G/T46G)
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Configuring Basic Features
Parameters Permitted Values
1 to 12 (for SIP-T42G)
1 to 6 (for SIP-T41P)
1-Line 1
2-Line 2
…
16-Line 16
Web User Interface:
DSSKey->Line Key X/ Programable Key->Line
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Account ID linekey.X.value/ programablekey.X.value
String within 99 characters
Default blank
Description:
Configures the group call pickup feature code.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Example: linekey.1.value = *98
Web User Interface:
DSSKey->Line Key X/ Programable Key->Value
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Value
To configure a group call pickup key via web user interface:
1. Click on DSSKey->Line Key X.
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 the 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.
3. Enter the group call pickup code in the Group Call Pickup Code field.
228
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.
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 GPickup from the Key Type field.
5. Press or , or the Switch soft key to select the desired line from the Account
ID field.
6. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
7. Enter the group call pickup code in the Value field.
8. Press the Save soft key to accept the change.
Call pickup is implemented through SIP signals on some specific servers. IP phones can pick up incoming calls via a NOTIFY message with dialog-info event. A user can pick up an incoming call by pressing a DSS key used to monitor a specific extension (such as a
BLF key).
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"
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Administrator’s Guide for SIP-T4X IP Phones 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 on the IP phone.
Parameter: account.X.dialoginfo_callpickup
Configure dialog-info call pickup on the IP phone.
Navigate to: http://<phoneIPAddress>/servlet?
p=account-adv&q=load&acc=0
Details of the Configuration Parameter:
Parameter Permitted Values Default account.X.dialoginfo_callpickup
0 or 1 0
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Configuring Basic Features
Parameter Permitted Values Default
Description:
Enables or disables the IP phone to pick up a call according to the SIP header of dialog-info for account X.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), call pickup is implemented through SIP signals.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->Dialog Info Call Pickup
Phone User Interface:
None
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.
4. Select the desired value from the pull-down list of Dialog Info Call Pickup.
5. Click Confirm to accept the change.
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232
Recall, also known as last call return, allows users to place a call back to the last caller.
Recall is implemented on IP phones using a recall key.
Procedure
Recall key can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Assign a recall key.
Parameter: linekey.X.type/ programablekey.X.type
Assign a recall key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&model=1&q=load&li nepage=1
Assign a recall key.
Recall Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameter
Permitted
Values
Default linekey.X.type/programablekey.X.type 7
Refer to the following content
Description:
Configures a DSS key as a recall key on the IP phone.
The digit 7 stands for the key type ReCall.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Example: linekey.1.type = 7
Configuring Basic Features
Parameter
Permitted
Values
Default
Default:
For line keys:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
For programable keys:
For
SIP-T48G/T46G
IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 (
Directory
).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61 (
Directory
).
When X=7, the default value is 51 (
Switch Account Up
).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=12, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
When X=14, the default value is 2 ( Forward ).
For SIP-T42G/T41P IP phones:
When X=1, the default value is 28 ( History ).
When X=2, the default value is 61 ( Directory ).
When X=3, the default value is 5 ( DND ).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 (
History
).
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Parameter
Permitted
Values
When X=6, the default value is 61 ( Directory ).
When X=7, the default value is 51 ( Switch Account Up ).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
Web User Interface:
DSSKey->Line Key X/ Programable Key->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type
Default
To configure a recall key via web user interface:
1. Click on DSSKey->Line Key X.
2. In the desired DSS key field, select ReCall from the pull-down list of Type.
234
3. Click Confirm to accept the change.
To configure a recall key via phone user interface:
1. Press Menu->Features->DSS Keys.
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select ReCall from the Key Type field.
5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Press the Save soft key to accept the change.
Configuring Basic Features
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.
Procedure
Call park key can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Phone User Interface
Assign a call park key. linekey.X.type
Assign a call park key.
Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&model=1&q=loa d&linepage=1
Assign a call park key.
Call Park Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameter Permitted Values Default linekey.X.type 10
Refer to the following content
Description:
Configures a line key to be a call park key on the IP phone.
The digit 10 stands for the key type Call Park.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.2.type = 10
Default:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29
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Parameter Permitted Values Default is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
Web User Interface:
DSSKey->Line Key X->Line Key X->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type linekey.X.line Integer from 1 to 16
1-6 correspond to the lines
1-6
Description:
Configures the desired line to apply the call park key.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
The valid values are:
1 to 16 (for SIP-T48G/T46G)
1 to 12 (for SIP-T42G)
1 to 6 (for SIP-T41P)
1-Line 1
2-Line 2
…
16-Line 16
Web User Interface:
DSSKey->Line Key X->Line Key X->Line
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Account ID linekey.X.value String Blank
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Configuring Basic Features
Parameter Permitted Values Default
Description:
Configures the call park feature code.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.2.value = *88
Web User Interface:
DSSKey->Line Key X->Line Key X->
Value
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Value
To configure a call park key via web user interface:
1. Click on DSSKey->Line Key X.
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.
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
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Administrator’s Guide for SIP-T4X IP Phones
ID field.
6. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
7. Enter the desired value (e.g., call park feature code) in the Value field.
8. Press the Save soft key to accept the change.
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 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 and stored in the call log.
For more information on calling line identification presentation, refer to
Calling and
Connected Line Identification Presentation on Yealink IP Phones
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
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.
Parameter: account.X.cid_source
Configure the presentation of the caller identity.
Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac c=0
Details of the Configuration Parameter:
Parameter Permitted Values Default account.X.cid_source 0, 1, 2, 3, 4 or 5 0
238
Configuring Basic Features
Parameter Permitted Values Default
Description:
Configures the presentation of the caller identity when receiving an incoming call 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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->Caller ID Source
Phone User Interface:
None
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.
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Administrator’s Guide for SIP-T4X IP Phones
4. Select the desired value from the pull-down list of the Caller ID Source.
240
5. Click Confirm to accept the change.
Connected line identification presentation (COLP) allows IP phones to display the identity of the connected party 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. Connected line identification presentation is also known as Called line identification presentation. In some cases, the remote party will be different from the called line identification presentation due to call diversion.
If the callee has existed in the local directory, the local contact name assigned to the callee should be preferentially displayed.
For more information on connected line identification presentation, refer to
Calling and
Connected Line Identification Presentation on Yealink IP Phones
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
Configuring Basic Features
Procedure
COLP can be configured only using the configuration files.
Configuration File <MAC>.cfg
Configure the presentation of the callee identity.
Parameter: account.X.cp_source
Details of the Configuration Parameter:
Parameter Permitted Values Default account.X.cp_source 0, 1 or 2 0
Description:
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 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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
None
Phone User Interface:
None
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
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Administrator’s Guide for SIP-T4X IP Phones 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
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 the 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 the 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 the 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 transmit
DTMF digits in three ways: DTMF, DTMF-Relay and Telephone-Event.
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Configuring Basic Features
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Configure the method of transmitting DTMF digit and the payload type.
Parameters: account.X.dtmf.type account.X.dtmf.dtmf_payload account.X.dtmf.info_type
Configure the number of times for the IP phone to send the end
RTP Event packet.
Parameter: features.dtmf.repetition
Configure the frequency level of DTMF digits.
Parameter: features.dtmf.volume
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
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.dtmf.type 0, 1, 2 or 3 1
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Parameters Permitted Values Default
Description:
Configures the DTMF type for account X.
0-INBAND
1-RFC 2833
2-SIP INFO
3-RFC2833 + SIP INFO
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 (RFC2833 + SIP INFO), DTMF digits are transmitted by RTP Events compliant to RFC 2833 and the SIP INFO messages.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->DTMF Type
Phone User Interface:
None account.X.dtmf.dtmf_payload Integer from 96 to 127 101
Description:
Configures the RFC 2833 payload type for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->DTMF Payload Type (96~127)
Phone User Interface:
None account.X.dtmf.info_type 1, 2 or 3 0
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Configuring Basic Features
Parameters Permitted Values Default
Description:
Configures the DTMF info type when the DTMF type is configured as “SIP INFO”,
“RFC2833 + SIP INFO” for account X.
0-Disabled
1-DTMF-Relay
2-DTMF
3-Telephone-Event
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->DTMF Info Type
Phone User Interface:
None features.dtmf.repetition 1, 2 or 3 3
Description:
Configures the repetition times for the IP phone to send the end RTP EVENT packet during an active call.
Web User Interface:
Features->General Information->DTMF Repetition
Phone User Interface:
None features.dtmf.volume Integer from -33 to 0 -10
Description:
Configures the frequency level of DTMF digits (in db).
Web User Interface:
None
Phone User Interface:
None
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.
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3. Click on Advanced.
4. Select the desired value from the pull-down list of DTMF Type.
If SIP INFO or RFC2833 + SIP INFO is selected, select the desired value from the pull-down list of DTMF Info Type.
5. Enter the desired value in the DTMF Payload Type (96~127) field.
6. 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.
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Configuring Basic Features
3. Click Confirm to accept the change.
Suppress DTMF display allows IP phones to suppress the display of DTMF digits. The 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.
Parameters: features.dtmf.hide features.dtmf.hide_delay
Configure suppress DTMF display and suppress DTMF display delay.
Navigate to: http://<phoneIPAddress>/servl et?p=features-general&q=loa d
Details of Configuration Parameters:
Parameters Permitted Values Default features.dtmf.hide 0 or 1 0
Description:
Enables or disables the IP phone to suppress the display of DTMF digits during an active call.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the DTMF digits are displayed as asterisks.
Web User Interface:
Features->General Information->Suppress DTMF Display
Phone User Interface:
None
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Parameters Permitted Values Default features.dtmf.hide_delay 0 or 1 0
Description:
Enables or disables the IP phone to display the DTMF digits for a short period before displaying asterisks during an active call.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter “features.dtmf.hide” is set to 1
(Enabled).
Web User Interface:
Features->General Information->Suppress DTMF Display Delay
Phone User Interface:
None
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.
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4. Click Confirm to accept the change.
Configuring Basic Features
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
Local
<y0000000000xx>.cfg
Web User Interface
Configure transfer via DTMF.
Parameters: features.dtmf.replace_tran features.dtmf.transfer
Configure transfer via DTMF.
Navigate to: http://<phoneIPAddress>/servl et?p=features-general&q=loa d
Details of Configuration Parameters:
Parameters Permitted Values Default features.dtmf.replace_tran 0 or 1 0
Description:
Enables or disables the IP phone to send DTMF sequences for transfer function when pressing the transfer soft key or the TRAN key.
0-Disabled
1-Enabled
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 designated DTMF digits to the server for completing call transfer when pressing the transfer key during a call.
Web User Interface:
Features->General Information->DTMF Replace Tran
Phone User Interface:
None features.dtmf.transfer
String within 32 characters
Blank
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Parameters Permitted Values Default
Description:
Configures the DTMF digits to be transmitted to perform call transfer.
Valid values are: 0-9, *, # and A-D.
Example: features.dtmf.transfer = 123
Note: It works only if the value of the parameter “features.dtmf.replace_tran” is set to
1 (Enabled).
Web User Interface:
Features->General Information->Tran Send DTMF
Phone User Interface:
None
To configure transfer via DTMF feature 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.
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Intercom allows establishing an audio conversation directly. The IP phone can answer intercom calls automatically. This feature depends on support from a SIP server.
Configuring Basic Features
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.
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.
Parameters: linekey.X.type linekey.X.line linekey.X.value
Assign an intercom key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&model=1&q=load&li nepage=1
Assign an intercom key.
Intercom Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters Permitted Values Default
Refer to the following content linekey.X.type 14
Description:
Configures a line key to be an intercom key.
The digit 14 stands for the key type Intercom.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.2.type = 14
Default:
For SIP-T48G IP phones:
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Parameters Permitted Values Default
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
Web User Interface:
DSSKey->Line Key X->Line KeyX->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type linekey.X.line Integer from 1 to 16
1-6 correspond to the lines 1-6
Description:
Configures the desired line to apply the intercom key.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
The valid values are:
1 to 16 (for SIP-T48G/T46G)
1 to 12 (for SIP-T42G)
1 to 6 (for SIP-T41P)
1-Line 1
2-Line 2
…
16-Line 16
Web User Interface:
DSSKey->Line Key X->Line KeyX->Line
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Account ID linekey.X.value String within 99 characters Blank
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Configuring Basic Features
Parameters Permitted Values Default
Description:
Configures the intercom number.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Web User Interface:
DSSKey->Line Key X->Line KeyX->Value
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Value
To configure an intercom key via web user interface:
1. Click on DSSKey->Line Key X.
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.
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. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Enter the remote extension number in the Value field.
7. 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 the incoming intercom call feature.
Parameters: features.intercom.allow features.intercom.mute features.intercom.tone features.intercom.barge
Configure the incoming intercom call feature.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-intercom&q=load
Configure the incoming intercom call feature.
Details of Configuration Parameters:
Parameters Permitted Values Default features.intercom.allow 0 or 1 1
Configuring Basic Features
Parameters Permitted Values Default
Description:
Enables or disables the IP phone to automatically answer an incoming intercom call.
0-Disabled
1-Enabled
If it is set to 0 (Disabled), the IP phone will reject incoming intercom calls and sends a busy signal to the caller.
If it is set to 1 (Enabled), the IP phone will automatically answer an incoming intercom call.
Web User Interface:
Features->Intercom ->Accept Intercom
Phone User Interface:
Menu->Features->Intercom ->Accept Intercom features.intercom.mute 0 or 1 0
Description:
Enables or disables the IP phone to mute the microphone when answering an intercom call.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the microphone is muted for intercom calls, and then the other party cannot hear you.
Note: It works only if the value of the parameter “ features.intercom.allow
” is set to 1
(Enabled).
Web User Interface:
Features->Intercom ->Intercom Mute
Phone User Interface:
Menu->Features->Intercom ->Intercom Mute features.intercom.tone 0 or 1 1
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Parameters Permitted Values Default
Description:
Enables or disables the IP phone to play a warning tone when receiving an intercom call.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter “ features.intercom.allow
” is set to 1
(Enabled).
Web User Interface:
Features->Intercom ->Intercom Tone
Phone User Interface:
Menu->Features->Intercom ->Intercom Tone features.intercom.barge 0 or 1 0
Description:
Enables or disables the IP phone to automatically answer an incoming intercom call while there is already an active call on the IP phone.
0-Disabled
1-Enabled
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.
Note: It works only if the value of the parameter “ features.intercom.allow
” is set to 1
(Enabled).
Web User Interface:
Features->Intercom->Intercom Barge
Phone User Interface:
Menu->Features->Intercom->Intercom Barge
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:
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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 or keyword parameter and maps it to the appropriate ring tone.
Note
If the caller already exists in the local directory, the ring tone assigned to the caller should be preferentially played.
Alert-Info headers in the following four formats:
Alert-Info: 127.0.0.1/Bellcore-drN (or Alert-Info: Bellcore-drN)
Alert-Info: ringtone-N (or Alert-Info: MyMelodyN)
Alert-Info: <URL>
Alert-Info: info=info text;x-line-id=0
When the Alter-Info header contains the keyword “Bellcore-drN”, the IP phone will play the Bellcore-drN (N=1, 2, 3, 4 or 5) ring tone if the parameter
“features.alert_info_tone” is set to 1, or play the corresponding local ring tone
(RingN.wav) in about ten seconds if the parameter “features.alert_info_tone” is set to 0.
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
Pattern
ID
1
2
3
Pattern Cadence
Ringing 2s On
Minimum
Duration
(ms)
1800
Nominal
Duration
(ms)
2000
Maximum
Duration
(ms)
2200
Silent 4s Off 3600 4000 4400
Ringing
Silent
Ringing
Silent
Long
Long
630
315
630
3475
800
400
800
4000
1025
525
1025
4400
Ringing
Silent
Ringing
Short
Short
315
145
315
400
200
400
525
525
525
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Configuring Advanced Features
Bellcore
Tone
Bellcore-dr4
Bellcore-dr5
Pattern
ID
4
5
Pattern Cadence
Silent
Minimum
Duration
(ms)
145
Nominal
Duration
(ms)
200
Maximum
Duration
(ms)
525
Ringing
Silent
Long 630
2975
800
4000
1025
4400
Ringing
Silent
Ringing
Silent
Short
Long
200
145
800
145
300
200
1000
200
525
525
1100
525
Ringing
Silent
Ringing
Short 200
2975
450
300
4000
500
525
4400
550
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.
When the Alter-Info header contains the keyword “ringtone-N” or “MyMolodyN”, the IP phone will play the corresponding local ring tone (RingN.wav), or play the first local ring tone (Ring1.wav) in about ten seconds if “N” is greater than 10 or less than 1.
Example:
Alert-Info: ringtone-2
Alert-Info: MyMelody2
The following table identifies the corresponding local ring tone:
Value of N
1
2
3
6
7
4
5
8
9
Ring Tone
Ring1.wav
Ring2.wav
Ring3.wav
Ring4.wav
Ring5.wav
Ring6.wav
Ring7.wav
Ring8.wav
Silent.wav
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Value of N
10
N<1 or N>10
Ring Tone
Splash.wav
Ring1.wav
When 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 (or the item called
“Distinctive Ring Tones” on the web user interface is Enabled), or play the preconfigured local ring tone in about ten seconds if the parameter
“account.X.alert_info_url_enable” is set to 0 or if the IP phone fails to download the remote ring tone.
Example:
Alert-Info: http://192.168.0.12:8080/Custom.wav
When the Alert-Info header contains an info text, the IP phone will map the text with the internal ringer text preconfigured on the IP phone, and then play the ring tone associated with the internal ringer text. If no internal ringer text maps, the IP phone will play the preconfigured local ring tone in about ten seconds.
Example:
Alert-Info: info=family;x-line-id=0
Auto Answer
If the Alert-Info header contains the following type of strings, the IP phone will answer incoming calls automatically without playing the ring tone:
Alert-Info: Auto Answer
Alert-Info: info = alert-autoanswer
Alert-Info: answer-after = 0 (or Alert-Info: Answer-After = 0)
Note
If the Alert-Info header contains multiple types of keywords, the IP phone will process the keywords in the following order：
AutoAnswer>URL>“Bellcore-drN/ringtone-N/MyMelodyN”>info text.
Procedure
Distinctive ring tones can be configured using the configuration files or locally.
Configuration File
<MAC>.cfg
<y0000000000xx>.cfg
Configure distinctive ring tones feature.
Parameter: account.X.alert_info_url_enable
Configure the internal ringer text and internal ringer file.
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Local Web User Interface
Configuring Advanced Features
Parameters: features.alert_info_tone distinctive_ring_tones.alert_info
.X.text distinctive_ring_tones.alert_info
.X.ringer
Configure distinctive ring tones feature.
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
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.alert_info_url_enable 0 or 1 1
Description:
Enables or disables the IP phone to download the ring tone from the URL contained in the Alert-Info header for account X.
0-Disabled
1-Enabled
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->Distinctive Ring Tones
Phone User Interface:
None features.alert_info_tone 0 or 1 0
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Parameters Permitted Values Default
Description:
Enables or disables the IP phone to map the keywords in the Alert-info header to the specified Bellcore ring tones.
0-Disabled
1-Enabled
Web User Interface:
None
Phone User Interface:
None distinctive_ring_tones.alert_info.X.text
(X ranges from 1 to 10)
String within 32 characters Blank
Description:
Configures the internal ringer text to map the keywords contained in the Alert-Info header.
Example: distinctive_ring_tones.alert_info.1.text = Family
Web User Interface:
Settings->Ring->Internal Ringer Text
Phone User Interface:
None distinctive_ring_tones.alert_info.X.ringer
(X ranges from 1 to 10)
Integer from 1 to 10 1
Description:
Configures the desired ring tones for each text.
The value ranges from 1 to 10, the digit stands for the appropriate ring tone.
1-Ring1.wav
2-Ring2.wav
3-Ring3.wav
4-Ring4.wav
5-Ring5.wav
6-Ring6.wav
7-Ring7.wav
8-Ring8.wav
9-Silent.wav
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Configuring Advanced Features
Parameters
10-Splash.wav
Web User Interface:
Settings->Ring->Internal Ringer File
Phone User Interface:
None
Permitted Values
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.
4. Select the desired value from the pull-down list of Distinctive Ring Tones.
Default
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.
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3. Select the desired ring tones for each text from the pull-down lists of Internal Ringer
File.
4. Click Confirm to accept the change.
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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:
Australia
Austria
Brazil
Belgium
China
Czech
Denmark
Finland
France
Germany
Great Britain
Greece
Hungary
Configuring Advanced Features
Lithuania
India
Italy
Japan
Mexico
New Zealand
Netherlands
Norway
Portugal
Spain
Switzerland
Sweden
Russia
United States
Chile
Czech ETSI
Configured tones can be heard on the IP phone 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-T48G IP phones.
When automatically answering a call Auto Answer
Procedure
Tones can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg Configure the tones for the IP
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Local Web User Interface phone.
Parameters: voice.tone.country 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
Configure the tones for the IP phone.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-tones&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default voice.tone.country Refer to the following content Custom
Description:
Configures the country tone for the IP phone.
Example: voice.tone.country = Custom
Permitted Values:
Custom, Australia, Austria, Brazil, Belgium, China, Czech, Denmark, Finland, France,
Germany, Great Britain, Greece, Hungary, Lithuania, India, Italy, Japan, Mexico,
New Zealand, Netherlands, Norway, Portugal, Spain, Switzerland, Sweden, Russia,
United States, Chile, Czech ETSI
Web User Interface:
Settings->Tones->Select Country
Phone User Interface:
None
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Configuring Advanced Features
Parameters Permitted Values Default voice.tone.dial String Blank
Description:
Customizes the dial tone. 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 0Hz, it means the tone is not played. A tone is comprised of at most four different frequencies.
Duration: the duration (in milliseconds) of the dial tone, ranges from 0 to 30000ms.
You can configure at most eight different tones for one condition, and separate them by commas. (e.g., 250/200, 0/1000, 200+300/500, 600+700+800+1000/2000).
If you want the IP phone to play tones once, add an exclamation mark “!” before tones (e.g., !250/200, 0/1000, 200+300/500, 600+700+800+1000/2000).
Note: It works only if the value of the parameter “voice.tone.country” is set to
Custom.
Web User Interface:
Settings->Tones->Dial
Phone User Interface:
None voice.tone.ring String Blank
Description:
Customizes the ringback tone.
The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
Note: It works only if the value of the parameter “voice.tone.country” is set to
Custom.
Web User Interface:
Settings->Tones->Ring Back
Phone User Interface:
None voice.tone.busy String Blank
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Parameters Permitted Values Default
Description:
Customizes the tone when the callee is busy.
The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
Note: It works only if the value of the parameter “voice.tone.country” is set to
Custom.
Web User Interface:
Settings->Tones->Busy
Phone User Interface:
None voice.tone.congestion String Blank
Description:
Customizes the tone when the network is congested.
The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
The default value is blank.
Note: It works only if the value of the parameter “voice.tone.country” is set to
Custom.
Web User Interface:
Settings->Tones->Congestion
Phone User Interface:
None voice.tone.callwaiting String Blank
Description:
Customizes the call waiting tone.
The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
The default value is blank.
Note: It works only if the value of the parameter “voice.tone.country” is set to
Custom.
Web User Interface:
Settings->Tones->Call Waiting
Phone User Interface:
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Configuring Advanced Features
Parameters
None voice.tone.dialrecall
Permitted Values Default
String Blank
Description:
Customizes the call back tone.
The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
Note: It works only if the value of the parameter “voice.tone.country” is set to
Custom.
Web User Interface:
Settings->Tones->Dial Recall
Phone User Interface:
None voice.tone.info String Blank
Description:
Customizes the info tone. The phone will play the info tone with the special information, for example, the number you are calling is not in service.
The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
The default value is blank.
Note: It works only if the value of the parameter “voice.tone.country” is set to
Custom.
Web User Interface:
Settings->Tones->Info
Phone User Interface:
None voice.tone.stutter String Blank
Description:
Customizes the tone when the IP phone receives a voice mail.
The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
The default value is blank.
Note: It works only if the value of the parameter “voice.tone.country” is set to
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Parameters
Custom.
Web User Interface:
Settings->Tones->Stutter
Phone User Interface:
None voice.tone.message
Permitted Values Default
String Blank
Description:
Customizes the tone when the IP phone receives a text message or voice message.
The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
The default value is blank.
Note: It works only if the value of the parameter “voice.tone.country” is set to
Custom.
Web User Interface:
Settings->Tones->Message
Phone User Interface:
None voice.tone.autoanswer String Blank
Description:
Customizes the warning tone for auto answer.
The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
The default value is blank.
Note: It works only if the value of the parameter “voice.tone.country” is set to
Custom.
Web User Interface:
Settings->Tones->Auto Answer
Phone User Interface:
None
To configure tones via web user interface:
1. Click on Settings->Tones.
2. Select the desired type from the pull-down list of Select Country.
If you select Custom, you can customize the tone for indicating each condition of the IP
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Configuring Advanced Features 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 entries, and then display the remote phone book entries on the phone user interface. IP phones support up to 5 remote phone books and 5000 entries. Remote phone book is customizable. For more
information, refer to Remote XML Phone Book on page 490 .
Sremote Name allows IP phones to search the entry names from the remote phone book for incoming/outgoing calls. Sremote Name Flash Time defines how often IP phones refresh the local cache of the remote phone book.
Procedure
Remote phone book can be configured using the configuration files or locally.
Configuration File
<y0000000000xx>.cf
g
Specify the access URL and the display name of the remote phone book.
Parameters: remote_phonebook.data.X.url remote_phonebook.data.X.name remote_phonebook.display_name
Specify whether to query the entry name from the remote phone book for outgoing/incoming calls.
Parameter: features.remote_phonebook.enable
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Local Web User Interface
Details of Configuration Parameters:
Specify how often the IP phone refreshes the local cache of the remote phone book.
Parameter: features.remote_phonebook.flash_tim
e
Specify whether to refresh the local cache of the remote phone book at a time when accessing the remote phone book.
Parameter: features.remote_phonebook.enter_up
date_enable
Specify the access URL of the remote phone book.
Navigate to: http://<phoneIPAddress>/servlet?p=c ontacts-remote&q=load
Specify whether to query the entry name from the remote phone book for outgoing/incoming calls.
Specify how often the IP phone refreshes the local cache of the remote phone book.
Navigate to: http://<phoneIPAddress>/servlet?p=c ontacts-remote&q=load
Parameters Permitted Values Default remote_phonebook.data.X.url
(X ranges from 1 to 5)
URL within 511 characters
Description:
Configures the access URL of the remote phone book.
Example: remote_phonebook.data.1.url = http://192.168.1.20/phonebook.xml
Web User Interface:
Blank
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Parameters
Directory->Remote Phone Book->Remote URL
Phone User Interface:
None
Permitted Values remote_phonebook.data.X.name
(X ranges from 1 to 5)
String within 99 characters
Description:
Configures the display name of the remote phone book item.
Example: remote_phonebook.data.1.name = Test
Web User Interface:
Directory->Remote Phone Book->Display Name
Phone User Interface:
None remote_phonebook.display_name
String within 99 characters
Default
Blank
Blank
Description:
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.
Example: remote_phonebook.display_name = Remote Phone Book
Note: It is not applicable to SIP-T42G/T41P IP phones.
Web User Interface:
None
Phone User Interface:
None features.remote_phonebook.enable
0 or 1 0
Description:
Enables or disables the IP phone to perform a remote phone book search for an incoming or outgoing call and display the matched call on the LCD screen.
0-Disabled
1-Enabled
Web User Interface:
Directory->Remote Phone Book->Incoming/Outgoing Call lookup
Phone User Interface:
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Parameters Permitted Values Default
None features.remote_phonebook.flash_time
0, Integer from
3600 to 2592000
21600
Description:
Configures how often to refresh the local cache of the remote phone book. If it is set to 3600, the IP phone will refresh the local cache of the remote phone book every
3600 seconds.
Note: If it is set to 0, the IP phone will not refresh the local cache of the remote phone book.
Web User Interface:
Directory->Remote Phone Book->Update Time Interval(Seconds)
Phone User Interface:
None features.remote_phonebook.enter_update_enable 0 or 1 0
Description:
Enables or disables the IP phone to refresh the local cache of the remote phone book at a time when accessing the remote phone book.
0-Disabled
1-Enabled
Web User Interface:
None
Phone User Interface:
None
To specify the 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.
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3. Enter the name in the Display Name field.
Configuring Advanced Features
4. Click Confirm to accept the change.
To configure the remote phone book via web user interface:
1. Click on Directory->Remote Phone Book.
2. Select the desired value from the pull-down list of Incoming/Outgoing Call lookup.
3. Enter the desired time in the Update Time Interval (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. The following LDAP servers are supported:
Microsoft Active Directory
Sun ONE Directory Server
Open LDAP Directory Server
Microsoft Active Directory Application Mode (ADAM)
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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 local directory.
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" may be used to select the desired entry or group, and return the desired information.
Configurations on the IP phone limit the amount of displayed entries when querying from the LDAP server, and decide how the attributes are displayed and sorted.
You can assign a DSS key to be an LDAP key, and press the LDAP key to enter the LDAP search screen when the IP phone is idle.
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
For more information on LDAP, refer to
LDAP Phonebook on Yealink IP Phones
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
Procedure
LDAP can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Configure the LDAP feature.
Parameters: ldap.enable ldap.name_filter
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Configuring Advanced Features
Local
Phone User Interface
Details of Configuration Parameters:
Parameters
Web User Interface ldap.number_filter ldap.tls_mode 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.call_out_lookup ldap.ldap_sort
Assign an LDAP key.
Parameter: linekey.X.type/ programablekey.X.type
Configure the LDAP feature.
Navigate to: http://<phoneIPAddress>/servl et?p=contacts-LDAP&q=load
Assign an LDAP key.
Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&model=1&q=loa d&linepage=1
Assign an LDAP key.
Permitted Values Default ldap.enable 0 or 1 0
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Parameters Permitted Values Default
Description:
Enables or disables LDAP feature on the IP phone.
0-Disabled
1-Enabled
Web User Interface:
Directory->LDAP->Enable LDAP
Phone User Interface:
None ldap.name_filter String within 99 characters Blank
Description:
Configures the criteria for searching the LDAP contact name attributes. 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.
Example: 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.
Web User Interface:
Directory->LDAP->LDAP Name Filter
Phone User Interface:
None ldap.number_filter String within 99 characters Blank
Description:
Configures the criteria for searching the LDAP contact number attributes. 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.
Example: ldap.number_filter = (|(telephoneNumber=%)(Mobile=%)(ipPhone=%))
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.
Web User Interface:
Directory->LDAP->LDAP Number Filter
Phone User Interface:
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Configuring Advanced Features
Parameters Permitted Values Default
None ldap.tls_mode Integer from 0 to 2 0
Description:
Configures the connection mode between the LDAP server and the IP phone.
0-LDAP—Unencrypted connection between LDAP server and the IP phone. (port 389 is used by default).
1-LDAP TLS Start—TLS/SSL connection between LDAP server and the IP phone (port
389 is used by default).
2-LDAPs—TLS/SSL connection between LDAP server and the IP phone (port 636 is used by default).
Web User Interface:
Directory->LDAP->LDAP TLS Mode
Phone User Interface:
None ldap.host String within 99 characters Blank
Description:
Configures the IP address or domain name of the LDAP server.
Example: ldap.host = 192.168.1.20
Web User Interface:
Directory->LDAP->Server Address
Phone User Interface:
None ldap.port Integer from 1 to 65535 389
Description:
Configures the port of the LDAP server.
Example: ldap.port = 389
Web User Interface:
Directory->LDAP->Port
Phone User Interface:
None
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Parameters Permitted Values Default ldap.base String within 99 characters Blank
Description:
Configures the LDAP search base which corresponds to the location of the LDAP phone book from which the LDAP search request begins. The search base narrows the search scope and decreases directory search time.
Example: ldap.base = dc=yealink,dc=cn
Web User Interface:
Directory->LDAP->Base
Phone User Interface:
None ldap.user String within 99 characters Blank
Description:
Configures the user name used 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 login the LDAP server.
Example: ldap.user = cn=manager,dc=yealink,dc=cn
Web User Interface:
Directory->LDAP->Username
Phone User Interface:
None ldap.password String within 99 characters Blank
Description:
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 login the LDAP server.
Example: ldap.password = secret
Web User Interface:
Directory->LDAP->Password
Phone User Interface:
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Configuring Advanced Features
Parameters
None ldap.max_hits
Permitted Values Default
Integer from 1 to 32000 50
Description:
Configures the maximum number of search results to be returned by the LDAP server.
If the value of the “Max.Hits” is blank, the 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.
Example: ldap.max_hits = 50
Web User Interface:
Directory->LDAP->Max. Hits (1~32000)
Phone User Interface:
None ldap.name_attr
String within 99 characters
Blank
Description:
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.
Example: ldap.name_attr = cn sn
Web User Interface:
Directory->LDAP->LDAP Name Attributes
Phone User Interface:
None ldap.numb_attr String within 99 characters Blank
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Parameters Permitted Values Default
Description:
Configures the number attributes of each record to be returned by the LDAP server.
You can configure multiple number attributes separated by spaces.
Example: ldap.numb_attr = telephoneNumber
Web User Interface:
Directory->LDAP->LDAP Number Attributes
Phone User Interface:
None ldap.display_name String within 99 characters Blank
Description:
Configures the display name of the contact record displayed on the LCD screen. The value must start with “%” symbol.
Example: ldap.display_name = %cn
The cn of the contact record is displayed on the LCD screen.
Web User Interface:
Directory->LDAP->LDAP Display Name
Phone User Interface:
None ldap.version 2 or 3 3
Description:
Configures the LDAP protocol version supported by the IP phone. Make sure the protocol value corresponds with the version assigned on the LDAP server.
Web User Interface:
Directory->LDAP->Protocol
Phone User Interface:
None ldap.call_in_lookup 0 or 1 0
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Configuring Advanced Features
Parameters Permitted Values Default
Description:
Enables or disables the IP phone to perform an LDAP search when receiving an incoming call.
0-Disabled
1-Enabled
Web User Interface:
Directory->LDAP->LDAP Lookup For Incoming Call
Phone User Interface:
None ldap.call_out_lookup 0 or 1 1
Description:
Enables or disables the IP phone to perform an LDAP search when placing a call.
0-Disabled
1-Enabled
Web User Interface:
Directory->LDAP->LDAP Lookup For Callout
Phone User Interface:
None ldap.ldap_sort 0 or 1 0
Description:
Enables or disables the IP phone to sort the search results in alphabetical order or numerical order.
0-Disabled
1-Enabled
Web User Interface:
Directory->LDAP->LDAP Sorting Results
Phone User Interface:
None
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LDAP Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameter
Permitted
Values
Default linekey.X.type/programablekey.X.type
38
Refer to the following content
Description:
Configures a DSS key as an LDAP key on the IP phone.
The digit 38 stands for the key type LDAP.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Example: linekey.2.type = 38
Default:
For line keys:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
For programable keys:
For
SIP-T48G/T46G
IP phones:
When X=1, the default value is 28 (
History
).
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Parameter
Permitted
Values
When X=2, the default value is 61 ( Directory ).
When X=3, the default value is 5 ( DND ).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 ( History ).
When X=6, the default value is 61 (
Directory
).
When X=7, the default value is 51 (
Switch Account Up
).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=12, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
When X=14, the default value is 2 (
Forward
).
For
SIP-T42G/T41P
IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 (
Directory
).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 ( History ).
When X=6, the default value is 61 ( Directory ).
When X=7, the default value is 51 ( Switch Account Up ).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
Web User Interface:
DSSKey->Line Key X
/ Programable Key
->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type
To configure LDAP via web user interface:
1. Click on Directory->LDAP.
2. Select Enabled from the pull-down list of Enable LDAP.
3. Enter the values in the corresponding fields.
Default
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4. Select the desired values from the corresponding pull-down lists.
5. Click Confirm to accept the change.
To configure an LDAP key via web user interface:
1. Click on DSSKey->Line Key X.
2. In the desired DSS key field, select LDAP from the pull-down list of Type.
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3. Click Confirm to accept the change.
To configure an LDAP key via phone user interface:
1. Press Menu->Features->DSS Keys.
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select LDAP from the Key Type field.
Configuring Advanced Features
5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Press the Save soft key to accept the change.
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 IP phone user status (busy or idle). Then when the 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 can press the BLF key to dial out the phone number. When the monitored user receives an incoming call, the supervisor can press the BLF key to pick up the call directly. When the monitored user is on a call, the supervisor can press the BLF key to interrupt and set up a conference call.
Visual Alert and Audio Alert for BLF Pickup
Visual alert 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 and 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 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
BLF LED Mode
BLF LED Mode provides four kinds of definition for the BLF key LED status. As there is no hard line key on SIP-T48G IP phones, BLF LED mode is only applicable to
SIP-T46G/T42G/T41P IP phones. BLF LED mode configuration is also applicable to the expansion module EXP40 connected to SIP-T48G/T46G IP phones. 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.
BLF LED mode feature is also applicable to BLF list key. For more information on BLF List
key, refer to BLF List on page 297 .
Line Key/Expansion Module Key LED (configured as a BLF key or a BLF List key and BLF
LED Mode is set to 0)
LED Status
Solid green
Fast flashing red (200ms)
Solid red
Description
The monitored user is idle.
The monitored user receives an incoming call.
The monitored user is dialing.
The monitored user is talking.
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Slow flashing red (1s)
Off
LED Status Description
The monitored user’s conversation is placed on hold (This LED status requires server support).
The call is parked against the monitored user’s phone number.
The monitored user does not exist.
Line Key/Expansion Module Key LED (configured as a BLF key or a BLF List key and BLF
LED Mode is set to 1)
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 (This LED status requires server support).
The call is parked against the monitored user’s phone number.
The monitored user is idle.
The monitored user does not exist.
Line Key/Expansion Module Key LED (configured as a BLF key a BLF List 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 monitored user’s conversation is placed on hold (This LED status requires server support).
The call is parked against the monitored user’s phone number.
The monitored user is idle.
The monitored user does not exist.
Line Key/Expansion Module Key LED (configured as a BLF key a BLF List key and BLF LED
Mode is set to 3)
LED Status
Fast flashing green (200ms)
Solid red
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 (This LED status requires server support).
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LED Status
Slow flashing red (1s)
Off
Description
The call is parked against the monitored user’s phone number.
The monitored user is idle.
The monitored user does not exist.
Note
BLF LED Mode feature is only applicable to IP phones running firmware version 72 or later.
Procedure
BLF can be configured using the configuration files or locally.
Configuration File
Local y0000000000xx.cfg
Web User Interface
Specify whether to use visual alert and audio alert for BLF pickup.
Parameters: features.pickup.blf_visual_enable features.pickup.blf_audio_enable
Assign a BLF key.
Parameters: linekey.X.type linekey.X.line linekey.X.value linekey.X.pickup_value
Configure BLF LED mode.
Parameter: features.blf_led_mode
Assign a BLF key.
Navigate to: http://<phoneIPAddress>/servlet?
p=dsskey&model=1&q=load&line page=1
Specify whether to use visual alert and audio alert for BLF pickup.
Navigate to: http://<phoneIPAddress>/servlet?
p=features-callpickup&q=load
Configure BLF LED mode.
Navigate to: http://<phoneIPAddress>/servlet?
p=features-general&q=load
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Phone User
Interface
Details of Configuration Parameters:
Assign a BLF key.
Parameters Permitted Values Default features.pickup.blf_visual_enable 0 or 1 0
Description:
Enables or disables the IP phone to display a visual alert when the monitored user receives an incoming call.
0-Disabled
1-Enabled
Web User Interface:
Features->Call Pickup->Visual Alert for BLF Pickup
Phone User Interface:
None features.pickup.blf_audio_enable 0 or 1 0
Description:
Enables or disables the IP phone to play an audio alert when the monitored user receives an incoming call.
0-Disabled
1-Enabled
Web User Interface:
Features->Call Pickup->Audio Alert for BLF Pickup
Phone User Interface:
None features.blf_led_mode 0, 1, 2 or 3 0
Description:
Configures BLF LED mode and provides four kinds of definition for the BLF key LED status.
Web User Interface:
Features->General Information->BLF LED Mode
Phone User Interface:
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Parameters Permitted Values Default
None
BLF Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters Permitted Values Default linekey.X.type 16
Refer to the following content
Description:
Configures a line key to be a BLF key on the IP phone.
The digit 16 stands for the key type BLF.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.2.type = 16
Default:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
Web User Interface:
DSSKey->Line Key X->Line KeyX->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type
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Parameters Permitted Values linekey.X.line Integer from 1 to 16
Description:
Configures the desired line to apply the BLF key.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
The valid values are:
1 to 16 (for SIP-T48G/T46G)
1 to 12 (for SIP-T42G)
1 to 6 (for SIP-T41P)
1-Line 1
2-Line 2
…
16-Line 16
Web User Interface:
DSSKey->Line Key X->Line KeyX->Line
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Account ID linekey.X.value String within 99 characters
Description:
Configures the number of the monitored user.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.3.value = 1008
Web User Interface:
DSSKey->Line Key X->Line KeyX->Value
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Value linekey.X.pickup_value
String within 256 characters
Default
1-6 correspond to the lines 1-6. blank
Blank
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Parameters Permitted Values Default
Description:
Configures the pickup code for BLF feature.
This parameter only applies to BLF feature.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.3.pickup_value = *97
Web User Interface:
DSSKey->Line Key X->Line KeyX->Extension
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->
Extension
To configure a BLF key via web user interface:
1. Click on DSSKey->Line Key X.
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 features 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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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.
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3. Click Confirm to accept the change.
To configure a BLF key via phone user interface:
1. Press Menu->Features->DSS Keys.
2. Select the desired DSS key.
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.
Configuring Advanced Features
5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Enter the phone number or extension you want to monitor in the Value field.
7. (Optional.) Enter the directed call pickup code in the Extension field.
8. Press the Save soft key to accept the change.
Busy Lamp Field (BLF) List allows a list of specific extensions to be monitored for status changes. It enables the monitoring phone to subscribe to a list of users, and receive notifications of the status of monitored users. Different indicators on the monitoring phone show the status of monitored users. The monitoring user can also be notified about calls being parked/no longer parked against any monitored user. IP phones support BLF list using a SUBSCRIBE/NOTIFY mechanism as specified in RFC 3265. This feature depends on support from a SIP server.
Note
BLF list feature is not applicable to SIP-T19P IP phones and applicable to IP phones running firmware version 73 or later in the neutral version.
Procedure
BLF List can be configured using the configuration files or locally.
Configuration File y0000000000xx.cfg
Configure BLF List.
Parameters: account.X.blf.blf_list_uri account.X.blf_list_code account.X.blf_list_barge_in_code account.X.blf_list_retrieve_call_parked_
code
Specify whether to automatically configure the BLF list keys.
Parameter: phone_setting.auto_blf_list_enable
Configure the order of BLF list keys assigned automatically.
Parameter: phone_setting.blf_list_sequence_type
Assign a BLF List key.
Parameters: linekey.X.type linekey.X.line
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Local
Web User Interface
Configure BLF List. http://<phoneIPAddress>/servlet?p=ac count-adv&q=load&acc=0
Assign a BLF List key.
Navigate to: http://<phoneIPAddress>/servlet?p=ds skey&model=1&q=load&linepage=1
Phone User
Interface
Assign a BLF List key.
Details of Configuration Parameters:
Default Parameters Permitted Values account.X.blf.blf_list_uri
String within 256 characters
Description:
Configures the BLF List URI to monitor a list of users for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.blf.blf_list_uri = [email protected]
Web User Interface:
Account->Advanced->BLF List URI
Phone User Interface:
None account.X.blf_list_code
String within 32 characters
Description:
Configures the directed pickup code for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.blf_list_code = *97
Note: It is not applicable to SIP-T19P IP phones.
Blank
Blank
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Configuring Advanced Features
Parameters Permitted Values
Web User Interface:
Account->Advanced->BLF List Code
Phone User Interface:
None account.X.blf_list_barge_in_code
String within 32 characters
Description:
Configures the barge-in code for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.blf_list_barge_in_code = *33
Note: It is not applicable to SIP-T19P IP phones.
Web User Interface:
Account->Advanced->BLF List Barge In Code
Phone User Interface:
None account.X.blf_list_retrieve_call_parked_code
String within 32 characters
Description:
Configures the call park retrieve code for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.blf_list_retrieve_call_parked_code = *88
Web User Interface:
Account->Advanced->BLF List Retrieve call parked Code
Phone User Interface:
None phone_setting.auto_blf_list_enable 0 or 1
Default
Blank
Blank
1
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Parameters Permitted Values Default
Description:
Enables or disables the IP phone to automatically configure the BLF list keys.
0-Disabled
1-Enabled
Web User Interface:
None
Phone User Interface:
None phone_setting.blf_list_sequence_type 0 or 1 0
Description:
Configures the order of BLF list keys assigned automatically.
0-Line Key->Ext Key
1-Ext Key->Line Key
Note: It is only applicable to SIP-T48G/T46G IP phones.
Web User Interface:
None
Phone User Interface:
None
BLF List Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters linekey.X.type
Permitted Values
39
Default
Refer to the following content
Description:
Configures a DSS key as a BLF List key on the IP phone.
The digit 39 stands for the key type BLF List.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
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Parameters Permitted Values Default
Example: linekey.2.type = 39
Default:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key
17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key
17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key
13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is 0.
Web User Interface:
DSSKey->Line Key X->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type linekey.X.line Integer from 1 to 16
1-6 correspond to the lines 1-6
Description:
Configures the desired line to apply the BLF List key.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
The valid values are:
1 to 16 (for SIP-T48G/T46G)
1 to 12 (for SIP-T42G)
1 to 6 (for SIP-T41P)
1-Line 1
2-Line 2
…
16-Line 16
Web User Interface:
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Parameters Permitted Values
DSSKey->
Line Key X/Programable Key
->Line
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Account ID
Default
To configure the BLF List settings 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 BLF List URI in the BLF List URL field.
5. (Optional.) Enter the directed pickup code in the BLF List Code field.
6. (Optional.) Enter the barge-in code in the BLF List Barge In Code field.
7. (Optional.) Enter the retrieve call parked code in the BLF List Retrieve call parked
Code field.
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8. Click Confirm to accept the change.
To configure BLF List keys manually via web user interface:
1. Click on DSSKey->Line Key X.
2. In the desired DSS key field, select BLF List from the pull-down list of Type.
3. Select the desired line from the pull-down list of Line.
Configuring Advanced Features
4. Repeat step 2-3, configure more BLF list keys.
5. Click Confirm to accept the change.
Hide Features Access Code feature enables the IP phone to display the feature identifier instead of the dialed feature access code automatically. For example, the dialed call park code will be replaced by the identifier “Call Park” when you park an active call.
The hide feature access codes feature is applicable to the following features:
Voice Mail
Pick up
Group Pick up
Barge In
Retrieve
Call Park
Group Park
Procedure
The hide feature access codes feature can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Configure the hide feature access codes feature:
Parameters: features.hide_feature_access_co
des.enable
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Local Web User Interface
Configure the hide feature access codes feature.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Details of Configuration Parameters:
Parameters
Permitted
Values
0 or 1
Default features.hide_feature_access_codes.enable 0
Description:
Enables or disables the IP phone to display feature name instead of the feature access code when dialing and in talk.
0-Disabled
1-Enabled
Web User Interface:
Features->General Information->Hide Feature Access Codes
Phone User Interface:
None
To enable hide feature access codes feature via web user interface:
1. Click on Features-> General Information.
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Configuring Advanced Features
2. Select Enabled from the pull-down list of Hide Feature Access Codes.
3. 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 IP 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 IP phone status and then decides whether to assign an incoming call to the user’s IP phone. When the IP 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 IP phone status or the ACD auto available timer (if configured) expires.
How long the IP phone remains unavailable is configurable by the auto available timer.
When the timer expires, the IP phone status is automatically changed to available. ACD auto available timer 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 on the IP phone indicates the ACD status.
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Procedure
ACD can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
<y0000000000xx>.cfg
Web User Interface
Configure ACD feature for account:
Parameters: account.X.acd.enable account.X.acd.available
Assign an ACD key.
Parameters: linekey.X.type
Configure ACD auto available.
Parameters: acd.auto_available acd.auto_available_timer
Assign an ACD key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&model=1&q=load&li nepage=1
Configure ACD auto available timer.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-acd&q=load
Assign an ACD key. Phone User Interface
Details of Configuration Parameters:
Parameters Permitted Values Default acd.auto_available
0 or 1
0
Description:
Enables or disables the IP phone to automatically change the status of the ACD agent to available after the designated time.
0-Disabled
1-Enabled
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Parameters
Web User Interface:
Features->ACD->ACD Auto Available
Phone User Interface:
None acd.auto_available_timer
Permitted Values Default
Integer from 0 to 120 60
Description:
Configures the length of time (in seconds) before the status of the ACD agent is automatically changed to available.
Web User Interface:
Features->ACD->ACD Auto Available Timer (0~120s)
Phone User Interface:
None account.X.acd.enable
0 or 1
0
Description:
Enables or disables ACD feature for account X.
0-Disabled
1-Enabled
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
None
Phone User Interface:
None account.X.acd.available 0 or 1 0
Description:
Enables or disables the IP phone to display the available and unavailable soft keys for account X after the IP phone logs into the ACD system.
0-Disabled
1-Enabled
X ranges from 1 to 16 (for SIP-T48G/T46G).
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Parameters
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
None
Phone User Interface:
None
Permitted Values Default
ACD Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameter Permitted Values Default linekey.X.type 42
Refer to the following content
Description:
Configures a line key to be an ACD key on the IP phone.
The digit 42 stands for the key type ACD.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.2.type = 42
Default:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
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Parameter Permitted Values
0.
Web User Interface:
DSSKey->Line Key X->Line KeyX->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type
Default
To configure an ACD key via web user interface:
1. Click on DSSKey->Line Key X.
2. In the desired DSS key field, select ACD from the pull-down list of Type.
3. Click Confirm to accept the change.
To configure the ACD auto available timer feature via web user interface:
1. Click on Features->ACD.
2. Select the desired value from the pull-down list of ACD Auto Available.
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3. Enter the desired timer in the 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.
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select ACD from the Type field.
4. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
5. Press the Save soft key to accept the change.
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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.
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 to message-summary updates. The server automatically sends a message-summary NOTIFY in a new dialog each time the MWI status changes.
Configuring Advanced Features
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure subscribe for MWI.
Parameters: account.X.subscribe_mwi account.X.subscribe_mwi_expires account.X.subscribe_mwi_to_vm
Configure subscribe MWI to voice mail.
Parameter: voice_mail.number.X
Configure the presentation of audio and visual MWI.
Parameter: account.X.display_mwi.enable
Configure subscribe for MWI.
Configure subscribe MWI to voice mail.
Configure the presentation of audio and visual MWI.
Navigate to: http://<phoneIPAddress>/servlet?p
=account-adv&q=load&acc=0
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.subscribe_mwi 0 or 1 0
Description:
Enables or disables the IP phone to subscribe the message waiting indicator 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.
0-Disabled
1-Enabled
X ranges from 1 to 16 (for SIP-T48G/T46G).
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Parameters
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->Subscribe for MWI
Phone User Interface:
None
Permitted Values account.X.subscribe_mwi_expires Integer from 0 to 84600
Default
3600
Description:
Configures MWI subscribe expiry time (in seconds) for account X.
The IP phone is able to successfully refresh the SUBSCRIBE for message-summary events before expiration of the SUBSCRIBE dialog.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Note: It works only if the value of the parameter “account.X.subscribe_mwi” is set to
1 (Enabled).
Web User Interface:
Account->Advanced->MWI Subscription Period (Seconds)
Phone User Interface:
None account.X.subscribe_mwi_to_vm 0 or 1 0
Description:
Enables or disables the IP phone to subscribe the message waiting indicator to the voice mail number for account X.
0-Disabled
1-Enabled
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Note: It works only if the value of the parameters “account.X.subscribe_mwi” is set to
1 (Enabled) and “voice_mail.number.X” is configured.
Web User Interface:
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Parameters Permitted Values
Account->Advanced->Subscribe MWI To Voice Mail
Phone User Interface:
None voice_mail.number.X
String within 99 characters
Default
Blank
Description:
Configures the voice mail number for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: voice_mail.number.1 = 1234
Note: It works only if the value of the parameter “account.X.subscribe_mwi_to_vm” is set to 1 (Enabled).
Web User Interface:
Account->Advanced->Voice Mail
Phone User Interface:
None account.X.display_mwi.enable
(X ranges from 1 to 6)
0 or 1 1
Description:
Enables or disables the IP phone to present audio and visual MWI when it receives new voice mails.
0-Disabled
1-Enabled
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->Voice Mail Display
Phone User Interface:
None
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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.
The IP phone will subscribe to the account number for MWI service by default.
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.
Configuring Advanced Features
5. Enter the desired voice number in the Voice Mail field.
6. Click Confirm to accept the change.
To configure the presentation of audio and visual 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 Voice Mail Display.
5. Click Confirm to accept the change.
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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.
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Users can send an RTP stream without involving SIP signaling by pressing a configured multicast paging key or paging list 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 that 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.
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Specify a multicast codec for the
IP phone to use for multicast RTP.
Parameter: multicast.codec
Assign a multicast paging key.
Parameters: linekey.X.type linekey.X.value
Assign a paging list key.
Parameters: linekey.X.type linekey.X.value
Configure the multicast IP address and port number for a paging list key.
Parameter: multicast.paging_address.X.ip_a
ddress
Configure the multicast paging group name for a paging list key.
Parameter:
Configuring Advanced Features
Local
Web User Interface
Phone User Interface multicast.paging_address.X.label
Assign a multicast paging key or a paging list key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&model=1&q=load&li nepage=1
Specify a multicast codec for the
IP phone to send the RTP stream.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Configure the multicast IP address and port number for a paging list key.
Configure the multicast paging group name for a paging list key.
Navigate to: http://<phoneIPAddress>/servlet
?p=contacts-multicastIP&q=load
Assign a multicast paging key or a paging list key.
Details of the Configuration Parameter:
Default Parameters Permitted Values multicast.codec
Refer to the following content
Description:
Configures the codec of multicast paging.
Example: multicast.codec = G722
Permitted Values:
PCMU, PCMA, G729, G722
Web User Interface:
Features->General Information->Multicast Codec
Phone User Interface:
None
G722
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Parameters Permitted Values Default multicast.paging_address.X.ip_address String Blank
Description:
Configures the multicast IP address and port number for a paging list key.
X ranges from 1 to 10.
Example: multicast.paging_address.1.ip_address = 224.5.6.20:10008
Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
Web User Interface:
Directory->Multicast IP->Paging List->Paging Address
Phone User Interface:
Menu->Features->Others->Option->Edit->Address multicast.paging_address.X.label String Blank
Description:
Configures the multicast paging group name for a paging list key.
X ranges from 1 to 10.
Example: multicast.paging_address.1.label = Product
Web User Interface:
Directory->Multicast IP->Paging List->Label
Phone User Interface:
Menu->Features->Others->Option->Edit->Label
Multicast Paging Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters Permitted Values Default linekey.X.type 24
Description:
Configures a line key to be a multicast paging key on the IP phone.
The digit 24 stands for the key type Multicast Paging.
Refer to the following content
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Configuring Advanced Features
Parameters Permitted Values Default
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.2.type = 24
Default:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
Web User Interface:
DSSKey->Line Key X->Line Key X->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type linekey.X.value
String within 99 characters
Blank
Description:
Configures the multicast IP address and port number.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
Web User Interface:
DSSKey->Line Key X->Line Key X->Value
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Value
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Paging List key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters
Permitted
Values
Default linekey.X.type/programablekey.X.type 66
Refer to the following content
Description:
Configures a DSS key as a paging list key on the IP phone.
The digit 66 stands for the key type Paging List.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Example: linekey.1.type = 66
Default:
For line keys:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
For programable keys:
For SIP-T48G/T46G IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 (
Directory
).
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Configuring Advanced Features
Parameters
Permitted
Values
When X=3, the default value is 5 ( DND ).
When X=4, the default value is 30 ( Menu ).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61 ( Directory ).
When X=7, the default value is 51 (
Switch Account Up
).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=12, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
When X=14, the default value is 2 (
Forward
).
For
SIP-T42G/T41P
IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61 (
Directory
).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61 ( Directory ).
When X=7, the default value is 51 ( Switch Account Up ).
When X=8, the default value is 52 ( Switch Account Down ).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 ( NA ).
When X=13, the default value is 0 (
NA
).
Web User Interface:
DSSKey->Line Key X/ Programable Key>Type
Phone User Interface:
Menu->Features->DSS Keys->Line Keys X->Type
Default
To configure a multicast paging key via web user interface:
1. Click on DSSKey->Line Key X.
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.
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The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
4. Click Confirm to accept the change.
To configure a paging list key via web user interface:
1. Click on DSSKey->Line Key X.
2. In the desired DSS key field, select Paging List from the pull-down list of Type.
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3. Click Confirm to accept the change.
To configure a codec for multicast paging via web user interface:
1. Click on Features ->General Information.
Configuring Advanced Features
2. Select the desired codec from the pull-down list of Multicast Codec.
3. Click Confirm to accept the change.
To configure multicast addresses for a paging list key via web user interface:
1. Click on Directory->Multicast IP.
2. Enter the multicast address and port number in the Paging Address field.
3. Enter the label in the Label field.
The label will appear on the LCD screen when sending the RTP multicast.
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4. Click Confirm to accept the change.
To configure a multicast paging key via phone user interface:
1. Press Menu->Features->DSS Keys.
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Paging from the Key Type field.
5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Enter the multicast IP address and port number in the Value field.
7. Press the Save soft key to accept the change.
To configure a paging list key via phone user interface:
1. Press Menu->Features->DSS Keys.
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Paging List from the Key Type field.
5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Press the Save soft key to accept the change.
To configure paging list via phone user interface:
1. Press Menu->Features->Others->Paging List.
2. Press the Option soft key.
3. Press the Edit soft key.
4. Enter the multicast IP address and port number (e.g., 224.5.6.20:10008) in the
Address field.
The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
5. Enter the group name in the Label field.
6. Press the Save soft key to accept the change.
Repeat the step 2-6, you can add more paging groups.
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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
Configuring Advanced Features will be automatically ignored. If the parameter is the priority value, the incoming multicast paging calls with higher priority are automatically answered and the ones with lower priority are ignored.
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
Web User Interface
Configure the listening multicast address.
Parameters: multicast.listen_address.X.label multicast.listen_address.X.ip_add
ress
Configure the Paging Barge and
Paging Priority Active features.
Parameters: multicast.receive_priority.enable multicast.receive_priority.priority
Configure the listening multicast address.
Configure the Paging Barge and
Paging Priority Active features.
Navigate to: http://<phoneIPAddress>/servlet
?p=contacts-multicastIP&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default multicast.listen_address.X.ip_address
(X ranges from 1 to 10)
IP address: port Blank
Description:
Configures the multicast address and port number that the IP phone listens to.
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Parameters Permitted Values Default
Example: multicast.listen_address.1.ip_address = 224.5.6.20:10008
Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
Web User Interface:
Directory->Multicast IP->Multicast Listening->Listening Address
Phone User Interface:
None multicast.listen_address.X.label
(X ranges from 1 to 10)
String within 99 characters Blank
Description:
Configures the label to be displayed on the LCD screen when receiving the RTP multicast.
Example: multicast.listen_address.1.label = Product
Web User Interface:
Directory->Multicast IP->Multicast Listening->Label
Phone User Interface:
None multicast.receive_priority.enable 0 or 1 1
Description:
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.
0-Disabled
1-Enabled
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.
Web User Interface:
Directory->Multicast IP->Multicast Listening->Paging Priority Active
Phone User Interface:
None multicast.receive_priority.priority Integer from 0 to 10 10
Description:
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Configuring Advanced Features
Parameters Permitted Values Default
Configures the priority of multicast paging calls.
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.
Web User Interface:
Directory->Multicast IP->Multicast Listening->Paging Barge
Phone User Interface:
None
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.
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 the 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.
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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.
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-T46G 28.72.0.1
Configuring Advanced Features
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
User-Agent: Yealink SIP-T46G 28.72.0.1
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-T46G 28.72.0.1 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:
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<YealinkIPPhoneText>
<Title>
</Title>
<Text>
Probably the recording box is full.
</Text>
<YealinkIPPhoneText>
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.
Parameters: linekey.X.type
Assign a URL record key.
Parameters: linekey.X.type linekey.X.value
Assign a record key and URL record key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&model=1&q=load&li nepage=1
Assign a record key and URL record key.
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Configuring Advanced Features
Record Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters Permitted Values Default linekey.X.type 25
Refer to the following content
Description:
Configures a line key to be a record key on the IP phone.
The digit 25 stands for the key type Record.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.1.type =25
Default:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
Web User Interface:
DSSKey->Line Key X->Line Key X->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type
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URL Record Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters Permitted Values Default linekey.X.type 35
Refer to the following content
Description:
Configures a line key to be a URL record key on the IP phone.
The digit 35 stands for the key type URL Record.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.1.type =35
Default:
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
Web User Interface:
DSSKey->Line Key X->Line KeyX->Type
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Type linekey.X.value
String within 99 characters
Blank
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Configuring Advanced Features
Parameters Permitted Values Default
Description:
Configures the URL to record a call.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
Example: linekey.2.value = http://10.1.2.224/phonerecording.cgi
Web User Interface:
DSSKey->Line Key X->Line KeyX->Value
Phone User Interface:
Menu->Features->DSS Keys->Line Key X->Value
To configure a record key via web user interface:
1. Click on DSSKey->Line Key X.
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->Line Key X.
2. In the desired DSS key field, select URL Record from the pull-down list of Type.
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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.
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. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Press the Save soft key to accept the change.
To configure a URL record key via phone user interface:
1. Press Menu->Features->DSS 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 URL Record field.
5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Press the Save soft key to accept the change.
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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 the 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. To use this feature, you need to assign a
Configuring Advanced Features 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.
Parameter: linekey.X.type/ programablekey.X.type
Assign a hot desking key.
Navigate to: http://<phoneIPAddress>/servlet
?p=dsskey&q=load&model=1
Phone User Interface Assign a hot desking key.
Hot Desking Key
For more information on how to configure the DSS Key, refer to Appendix C: Configuring
on page
Parameters
Permitted
Values
Default linekey.X.type/programablekey.X.type 34
Refer to the following content
Description:
Configures a DSS key as a hot desking key on the IP phone.
The digit 34 stands for the key type Hot Desking.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
Example: linekey.2.type = 34
Default:
For line keys:
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Parameters
Permitted
Values
Default
For SIP-T48G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
For programable keys:
For SIP-T48G/T46G IP phones:
When X=1, the default value is 28 ( History ).
When X=2, the default value is 61 (
Directory
).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61 (
Directory
).
When X=7, the default value is 51 (
Switch Account Up
).
When X=8, the default value is 52 (
Switch Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=12, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
When X=14, the default value is 2 (
Forward
).
For
SIP-T42G/T41P
IP phones:
When X=1, the default value is 28 ( History ).
When X=2, the default value is 61 ( Directory ).
When X=3, the default value is 5 ( DND ).
When X=4, the default value is 30 ( Menu ).
When X=5, the default value is 28 ( History ).
When X=6, the default value is 61 (
Directory
).
When X=7, the default value is 51 (
Switch Account Up
).
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Configuring Advanced Features
Parameters
Permitted
Values
When X=8, the default value is 52 ( Switch Account Down ).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 (
NA
).
When X=13, the default value is 0 ( NA ).
Web User Interface:
DSSKey->Line Key X
/Programable Key
->Type
Phone User Interface:
None
Default
To configure a hot desking key via web user interface:
1. Click on DSSKey->Line Key X.
2. In the desired DSS key field, select Hot Desking from the pull-down list of Type.
3. Click Confirm to accept the change.
To configure a hot desking key via phone user interface:
1. Press Menu->Features->DSS Keys.
2. Select the desired DSS key.
3. Press or , or the Switch soft key to select Key Event from the Type field.
4. Press or , or the Switch soft key to select Hot Desking from the Key Type field.
5. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
6. Press the Save soft key to accept the change.
Action URL allows IP phones to interact with web server applications by sending an
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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
Description
When the IP phone completes startup.
When the IP phone successfully registers an account.
Unregistered
Register Failed
Off Hook
On Hook
Incoming Call
Outgoing Call
Established
Terminated
Open DND
Close DND
Open Always Forward
Close Always Forward
Open Busy Forward
Close Busy Forward
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.
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.
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
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 phone changes.
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Event
Forward Incoming Call
Reject Incoming Call
Answer New-In Call
Transfer Finished
Transfer Failed
Idle to Busy
Busy to Idle
Autop Finish
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=$variable value. Variable name can be customized by users, while the variable value is pre-defined. For example, a URL “
http://192.168.1.10/help.xml?mac=$mac
” is specified for the event Mute, $mac will be dynamically replaced with the MAC address of the phone when the IP phone mutes a call.
The following table lists the pre-defined variable values.
$mac
Variable Value
$ip
$model
$firmware
Description
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.
When the IP phone completes auto provisioning via power on.
$active_url
$active_user
$active_host
$local
Description
The MAC address of the phone
The IP address of the 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
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Variable Value
$remote
$display_local
$display_remote
Description 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.
The call-id of the active call. $call_id
Procedure
Action URL can be configured using the configuration files or locally.
Configuration File
<y0000000000xx>.cf
g
Configure the action URL.
Parameters: 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
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Local Web User Interface 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 action_url.transfer_failed action_url.setup_autop_finish
Configure the action URL.
Navigate to: http://<phoneIPAddress>/servlet?p=f eatures-actionurl&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default action_url.setup_completed URL within 511 characters
Description:
Configures the action URL the IP phone sends after startup.
The value format is: http(s)://IP address of server/help.xml? variable name=variable value.
Valid variable values are:
$mac
$ip
$model
Blank
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Parameters Permitted Values
$firmware
$active_url
$active_user
$active_host
$local
$remote
$display_local
$display_remote
$call_id
Example: action_url. setup_completed = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Setup Completed action_url.registered
URL within 511 characters
Default
Blank
Description:
Configures the action URL the IP phone sends after an account is registered.
Example: action_url.registered = http://192.168.0.20/help.xml?IP=$ip
Note: The old parameter “action_url.log_on” is also applicable to IP phones.
Web User Interface:
Features->Action URL->Registered
Phone User Interface:
None action_url.unregistered
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends after an account is unregistered.
Example: action_url.unregistered = http://192.168.0.20/help.xml?IP=$ip
Note: The old parameter “action_url.log_off” is also applicable to IP phones.
Web User Interface:
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Configuring Advanced Features
Parameters
Features->Action URL->Unregistered
Phone User Interface:
None action_url.register_failed
Permitted Values Default
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends after a register failed.
Example: action_url.register_failed = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Register Failed
Phone User Interface:
None action_url.off_hook
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when off hook.
Example: action_url.off_hook = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Off Hook
Phone User Interface:
None action_url.on_hook
URL within 511 characters
Description:
Configures the action URL the IP phone sends when on hook.
Example: action_url.on_hook = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->On Hook
Blank
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Parameters
Phone User Interface:
None action_url.incoming_call
Permitted Values Default
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when receiving an incoming call.
Example: action_url.incoming_call = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Incoming Call
Phone User Interface:
None action_url.outgoing_call
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when placing a call.
Example: action_url.outgoing_call = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Outgoing Call
Phone User Interface:
None action_url.call_established
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when establishing a call.
Example: action_url.call_established = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Established
Phone User Interface:
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Configuring Advanced Features
Parameters Permitted Values Default
None action_url.dnd_on
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when DND feature is enabled.
Example: action_url.dnd_on = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Open DND
Phone User Interface:
None action_url.dnd_off
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when DND feature is disabled.
Example: action_url.dnd_off = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Close DND
Phone User Interface:
None action_url.always_fwd_on
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when always forward feature is enabled.
Example: action_url.always_fwd_on = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Open Always Forward
Phone User Interface:
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Parameters
None action_url.always_fwd_off
Permitted Values Default
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when always forward feature is disabled.
Example: action_url.always_fwd_off = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Close Always Forward
Phone User Interface:
None action_url.busy_fwd_on
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when busy forward feature is enabled.
Example: action_url.busy_fwd_on = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Open Busy Forward
Phone User Interface:
None action_url.busy_fwd_off
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when busy forward feature is disabled.
Example: action_url.busy_fwd_off = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Close Busy Forward
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Configuring Advanced Features
Parameters
Phone User Interface:
None action_url.no_answer_fwd_on
Permitted Values Default
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when no answer forward feature is enabled.
Example: action_url.no_answer_fwd_on = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Open No Answer Forward
Phone User Interface:
None action_url.no_answer_fwd_off
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when no answer forward feature is disabled.
Example: action_url.no_answer_fwd_off = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Close No Answer Forward
Phone User Interface:
None action_url.transfer_call
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when performing a transfer.
Example: action_url.transfer_call = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Transfer Call
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Parameters
Phone User Interface:
None action_url.blind_transfer_call
Permitted Values Default
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when performing a blind transfer.
Example: action_url.blind_transfer_call = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Blind Transfer
Phone User Interface:
None action_url.attended_transfer_call
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when performing an attended/semi-attended transfer.
Example: action_url.attended_transfer_call = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Attended Transfer
Phone User Interface:
None action_url.hold
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when placing a call on hold.
Example: action_url.hold = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Hold
Phone User Interface:
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Configuring Advanced Features
Parameters
None action_url.unhold
Permitted Values Default
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when resuming a held call.
Example: action_url.unhold = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->UnHold
Phone User Interface:
None action_url.mute
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when muting a call.
Example: action_url.mute = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Mute
Phone User Interface:
None action_url.unmute
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when un-muting a call.
Example: action_url.unmute = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->UnMute
Phone User Interface:
None
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Parameters Permitted Values Default action_url.missed_call
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when missing a call.
Example: action_url.missed_call = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Missed Call
Phone User Interface:
None action_url.call_terminated
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when terminating a call.
Example: action_url.call_terminated = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Terminated
Phone User Interface:
None action_url.busy_to_idle
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when changing the state of the IP phone from busy to idle.
Example: action_url.busy_to_idle = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Busy To Idle
Phone User Interface:
None
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Configuring Advanced Features
Parameters Permitted Values Default action_url.idle_to_busy
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when changing the state of the IP phone from idle to busy.
Example: action_url.idle_to_busy = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Idle To Busy
Phone User Interface:
None action_url.ip_change
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when changing the IP address of the
IP phone.
Example: action_url.ip_change = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->IP Changed
Phone User Interface:
None action_url.forward_incoming_call
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when forwarding an incoming call.
Example: action_url.forward_incoming_call = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Forward Incoming Call
Phone User Interface:
None
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Parameters Permitted Values Default action_url.reject_incoming_call
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when rejecting an incoming call.
Example: action_url.reject_incoming_call = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Reject Incoming Call
Phone User Interface:
None action_url.answer_new_incoming_call
URL within 511 characters
Blank
Description:
Configures the action URL the IP phone sends when answering a new incoming call.
Example: action_url.answer_new_incoming_call = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Answer New-In Call
Phone User Interface:
None action_url.transfer_finished
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when completing a call transfer.
Example: action_url.transfer_finished = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Transfer Finished
Phone User Interface:
None action_url.transfer_failed
URL within 511 characters Blank
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Configuring Advanced Features
Parameters Permitted Values Default
Description:
Configures the action URL the IP phone sends when failing to transfer a call.
Example: action_url.transfer_failed = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Transfer Failed
Phone User Interface:
None action_url.setup_autop_finish
URL within 511 characters Blank
Description:
Configures the action URL the IP phone sends when completing auto provisioning via power on.
Example: action_url.setup_autop_finish = http://192.168.0.20/help.xml?IP=$ip
Web User Interface:
Features->Action URL->Autop Finish
Phone User Interface:
None
To configure action URL via web user interface:
1. Click on Features->Action URL.
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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, which are separated by “=”. The valid URI format is: http(s)://phone IP address/servlet?key=variable value.
The following table lists the pre-defined variable values.
Variable Value
OK
ENTER
SPEAKER
F_TRANSFER
VOLUME_UP
VOLUME_DOWN
MUTE
F_HOLD
Phone Action
Press the OK key.
Press the Enter soft key
Press the Speakerphone key.
Transfers a call to another party.
Increase the volume.
Decrease the volume.
Mute a call.
Place an active call on hold.
Variable Value
X
CANCEL
0-9/*/POUND
L1-LX
F_CONFERENCE
F1-F4
MSG
HEADSET
RD
UP/DOWN/LEFT/RIGHT
Reboot
AutoP
DNDOn
DNDOff number=xxx&outgoing_uri=y
OFFHOOK
ONHOOK
ANSWER
Reset
ATrans=xxx
BTrans=xxx
CALLEND
Configuring Advanced Features
Phone Action
Cancel actions or reject incoming calls.
Return to a previous screen or cancel a call.
Press the keypad (0-9, * or #).
Press the line key (for SIP-T48G, X=29,
SIP-T46G, X=27, for SIP-T42G/T41P, X=15).
Press the Conference soft key.
Press the soft keys.
Press the MESSAGE key.
Press the HEADSET key.
Press the REDIAL key.
Press the navigation keys.
Reboot the phone.
Perform auto provisioning.
Activate the DND mode.
Deactivate the DND mode.
Place a call to xxx from SIP URI y.
Pick up the handset.
Hang up the handset.
Answer a call.
Reset a phone.
Perform a semi-attended/attended transfer
End a call.
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Variable Value Phone Action 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].
Example: http://10.3.20.10/servlet?phonecfg=get[&acc ounts=1][&dnd=0][&fw=1]
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 specified 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. You can use action URI feature to capture the phone’s current screen.
Procedure
Specify the trusted IP address for Action URI using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Specify the trusted IP address(es) for sending the
Action URI to the IP phone.
Parameter: features.action_uri_limit_ip
Specify the trusted IP address(es) for sending the
Action URI to the IP phone.
Navigate to: http://<phoneIPAddress>/servl
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Configuring Advanced Features et?p=features-remotecontrl&q
=load
Details of the Configuration Parameter:
Parameter Permitted Values Default features.action_uri_limit_ip IP address or any Blank
Description:
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.
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 will reject any HTTP GET request.
If it is set to “any”, the IP phone will accept and handle HTTP GET requests from any
IP address.
Example: features.action_uri_limit_ip = 10.3.6.117,10.3.6.119
Web User Interface:
Features->Remote Control->Action URI allow IP List
Phone User Interface:
None
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.
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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.
You can capture the screen display of the IP phone using the action URI. IP phones support handling an HTTP or HTTPS GET request. The URI format is http(s)://<phoneIPAddress>/screencapture. The captured picture can be saved as a
BMP or JPEG file.
You can also use the URI “http(s)://<phoneIPAddress>/screencapture/download” to capture the screen display first, and then download the image (which is saved as a JPG file and named with the phone model and the capture time) to the local system. Before capturing the phone’s current screen, ensure that the IP address of the computer is included in the trusted IP address for Action URI on the phone.
When you capture the screen display, the IP phone may prompt you to enter the user name and password of the administrator if web browser does not remember the user name and password for web user interface login.
Note
IP phones also support capturing the screen display using the old URI
“ http://<phoneIPAddress>/servlet?command=screenshot
”.
To capture the current screen of the phone:
1. Enter request URI (e.g., http://10.3.6.104/screencapture) in the browser's address
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Configuring Advanced Features bar and press the Enter key on the keyboard.
2. Do one of the following:
-
If it is the first time you capture the phone’s current screen using the computer, the browser will display “remote control forbidden”, and the LCD screen will prompt the message “Allow Remote Control?”.
Press the OK soft key on the phone to allow remote control. The phone will return to the previous screen.
Refresh the web page.
The browser will display an image showing the phone’s current screen. You can then save the image to your local system.
- Else, the browser will display an image showing the phone’s current screen display directly. You can save the image to your local system.
Note
Frequent capture may affect the phone performance. Yealink recommend you to capture the phone screen display within a minimum interval of 4 seconds.
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
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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 mechanisms from the primary to the secondary server.
Fallback: In this mode, a second less featured call server (fallback server) with SIP capability takes over call control to provide basic calling capability, but without some advanced features (for example, shared lines, call recording and MWI) offered by the working server. 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.
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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 physical SIP servers for failover purpose. The working server is deployed in redundant pairs, designated as primary and secondary servers. The primary server is 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 address of the fallback server. For example, 192.168.1.15. A fallback server offers less functionality than the working server.
Configuring Advanced Features
Phone Registration
Two registration methods for fallback mode:
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.
For more information on server redundancy, refer to
Server Redundancy on Yealink IP
Phones
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
Procedure
Server redundancy can be configured using the configuration files or locally.
Configuration File <MAC>.cfg
Local
Web User
Interface
Configure the server redundancy on the IP phone.
Parameters: account.X.sip_server.Y.address account.X.sip_server.Y.port account.X.sip_server.Y.expires account.X.sip_server.Y.retry_counts
Fallback Mode: account.X.fallback.redundancy_type account.X.fallback.timeout
Failover Mode: account.X.sip_server.Y.failback_mode account.X.sip_server.Y.failback_timeout account.X.sip_server.Y.register_on_enable
Configure the server redundancy on the IP phone.
Navigate to: http://<phoneIPAddress>/servlet?p=account
-register&q=load&acc=0
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Details of Configuration Parameters:
Parameters Permitted Values Default account.X.sip_server.Y.address
(X ranges from 1 to 16. Y ranges from 1 to 2)
String within 256 characters
Blank
Description:
Configures the IP address or domain name of the SIP server Y for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.sip_server.1.address = yealink.pbx.com
Web User Interface:
Account->Register ->SIP Server Y->Server Host
Phone User Interface:
None account.X.sip_server.Y.port
(X ranges from 1 to 16. Y ranges from 1 to 2)
Integer from 0 to 65535 5060
Description:
Configures the port of the SIP server Y for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.sip_server.1.port = 5060
Web User Interface:
Account->Register ->SIP Server Y->Port
Phone User Interface:
None account.X.sip_server.Y.expires
(X ranges from 1 to 16. Y ranges from 1 to 2)
Integer from 30 to
2147483647
3600
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Parameters Permitted Values Default
Description:
Configures the registration expiration time (in seconds) of the SIP server Y for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.sip_server.1.expires = 3600
Web User Interface:
Account->Register ->SIP Server Y->Server Expires
Phone User Interface:
None account.X.sip_server.Y.retry_counts
(X ranges from 1 to 16. Y ranges from 1 to 2)
Integer from 0 to 20 3
Description:
Configures the retry times for the IP phone to resend requests when the SIP server Y is unavailable or there is no response from the SIP server Y for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Register->SIP Server Y ->Server Retry Counts
Phone User Interface:
None account.X.fallback.redundancy_type
(X ranges from 1 to 16)
0 or 1 0
Description:
Configures the registration mode for the IP phone in fallback mode.
0-Concurrent Registration
1-Successive Registration
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
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Parameters Permitted Values Default
Web User Interface:
None
Phone User Interface:
None account.X.fallback.timeout
Integer from 10 to
2147483647
120
Description:
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 the Successive Registration mode.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
None
Phone User Interface:
None account.X.sip_server.Y.failback_mode
(X ranges from 1 to 16, Y ranges from 1 to 2)
0, 1, 2 or 3 0
Description:
Configures the way in which the phone fails back to the primary server for call control in the failover mode.
0-newRequests: all requests are sent to the primary server first, regardless of the last server that was used.
1-DNSTTL: the IP phone will send requests to the last registered server first. If the time defined by DNSTTL on the registered server expires, the phone will retry to send requests to the primary server.
2-registration: the IP phone will send requests to the last registered server first. If the registration expires, the phone will retry to send requests to the primary server.
3-duration: the IP phone will send requests to the last registered server first. If the time defined by the account.X.sip_server.Y.failback_timeout parameter expires, the phone will retry to send requests to the primary server.
X ranges from 1 to 16 (for SIP-T48G/T46G).
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Parameters
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
None
Phone User Interface:
None account.X.sip_server.Y.failback_timeout
(X ranges from 1 to 16, Y ranges from 1 to 2)
Permitted Values
0, 60 to 65535
Default
3600
Description:
Configures the time (in seconds) for the 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 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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
None
Phone User Interface:
None account.X.sip_server.Y.register_on_enable
(X ranges from 1 to 16, Y ranges from 1 to 2)
0 or 1 0
Description:
Enables or disables the IP phone to register to the secondary server when sending requests to the secondary server in the failover mode.
0-Disabled
1-Enabled
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
None
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Parameters Permitted Values
Phone User Interface:
None
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. Select the desired value from the pull-down list of Transport.
4. Configure parameters of the SIP server 1 in the corresponding fields.
5. Configure parameters of the SIP server 2 in the corresponding fields.
Default
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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.
5. Configure parameters of the SIP server 1 or SIP server 2 in the corresponding fields.
Configuring Advanced Features
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 discovered 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 the NAPTR query for the SRV 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, 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.
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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.
NAPTR (Naming Authority Pointer)
First, the IP phone sends the NAPTR query to get the SRV pointer and 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.
The flag “s” means to perform an SRV lookup. flags service
Specify the transport protocols supported by the domain server:
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 the NAPTR query again according to the previous NAPTR query result.
SRV (Service Location Record)
The IP phone performs a 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 the
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 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 IP 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
SIP Server Domain Name Resolution 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.
Parameters: account.X.transport account.X.naptr_build
Configure the transport type on the IP phone.
Navigate to: http://<phoneIPAddress>/se rvlet?p=account-register&q
=load&acc=0
Details of Configuration Parameters:
Parameters Permitted Values Default account.X.transport 0, 1, 2 or 3 0
Description:
Configures the type of transport protocol for account X.
0-UDP
1-TCP
2-TLS
3-DNS-NAPTR
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Register ->Transport
Phone User Interface:
None
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Parameters Permitted Values Default account.X.naptr_build 0 or 1 0
Description:
Configures the way of SRV query for the IP phone to be performed when no result is returned from NAPTR query for account X.
0-SRV query using UDP only
1-SRV query using UDP, TCP and TLS
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
None
Phone User Interface:
None
Failover redundancy can only be utilized when the configured domain name of the SIP server is resolved to multiple IP addresses. If the IP phone is not configured with a DNS server, or the DNS query returns no result from a DNS server, you can configure a set of
DNS NAPTR/SRV/A records into the IP phone. The IP phone will attempt to resolve the domain name of the SIP server with static DNS cache.
When the IP phone is configured with a DNS server, the IP phone will behave as follows to resolve domain name of the SIP server:
The IP phone performs a DNS query to resolve the domain name from the DNS server.
If the DNS query returns no results for the domain name, or the returned record cannot be contacted, the values in the static DNS cache (if configured) are used when their configured time intervals are not elapsed.
If the configured time interval is elapsed, the IP phone will attempt to perform a
DNS query again.
If the DNS query returns a result, the IP phone will use the returned record and ignore the statically configured cache values.
When the IP phone is not configured with a DNS server, it will behave as follow:
The IP phone attempts to resolve the domain name within the static DNS cache.
The IP phone will always use the results returned from the static DNS cache.
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IP phones can be configured to use static DNS cache preferentially. Static DNS cache is configurable on a per-line basis.
Procedure
Static DNS cache can be configured only using the configuration files.
Configuration File
<y0000000000 xx>.cfg
<MAC>.cfg
Configure NAPTR/SRV/A records.
Parameters: dns_cache_naptr.X.name dns_cache_naptr.X.flags dns_cache_naptr.X.order dns_cache_naptr.X.preference dns_cache_naptr.X.replace dns_cache_naptr.X.service dns_cache_naptr.X.ttl dns_cache_srv.X.name dns_cache_srv.X.port dns_cache_srv.X.priority dns_cache_srv.X.target dns_cache_srv.X.weight dns_cache_srv.X.ttl dns_cache_a.X.name dns_cache_a.X.ip dns_cache_a.X.ttl
Configure the IP phone whether to cache the additional DNS records.
Parameter: account.X.dns_cache_type
Configure the IP phone whether to use static DNS cache preferentially.
Parameter: account.X.static_cache_pri
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Details of Configuration Parameters:
Parameters Permitted Values Default dns_cache_naptr.X.name
(X ranges from 1 to 12)
String within 256 characters
Description:
Configures the domain name to which NAPTR record X refers.
Example: dns_cache_naptr.1.name = yealink.pbx.com
Web User Interface:
None
Phone User Interface:
None dns_cache_naptr.X.flags
(X ranges from 1 to 12)
S, A, U or P
Blank
Blank
Description:
Configures the flag of NAPTR record X. (Always “s” for SIP, which means to do an
SRV lookup on whatever is in the replacement field).
S-Do an SRV lookup next.
A-Do an A lookup next.
U-No need to do a DNS query next.
P-Service customized by the user
Example: dns_cache_naptr.1.flags = S
Web User Interface:
None
Phone User Interface:
None dns_cache_naptr.X.order
(X ranges from 1 to 12)
Integer from 0 to 65535 0
Description:
Configures the order of NAPTR record X.
NAPTR record with lower order is more preferred.
Example:
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Parameters dns_cache_naptr.1.order = 90
Web User Interface:
None
Phone User Interface:
None dns_cache_naptr.X.preference
(X ranges from 1 to 12)
Permitted Values
Integer from 0 to 65535
Default
0
Description:
Configures the preference of NAPTR record X. NAPTR record with lower preference is more preferred.
Example: dns_cache_naptr.1.preference = 50
Web User Interface:
None
Phone User Interface:
None dns_cache_naptr.X.replace
(X ranges from 1 to 12)
Domain name Blank
Description:
Configures a domain name to be used for the next SRV query in NAPTR record X.
Example: dns_cache_naptr.1.replace = _sip._tcp.yealink.pbx.com
Web User Interface:
None
Phone User Interface:
None dns_cache_naptr.X.service
(X ranges from 1 to 12)
String within 32 characters
Blank
Description:
Configures the transport protocol available for the SIP server in NAPTR record X.
SIP+D2U: SIP over UDP
SIP+D2T: SIP over TCP
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Parameters
SIP+D2S: SIP over SCTP
SIPS+D2T: SIPS over TCP
Example: dns_cache_naptr.1.service = SIP+D2T
Web User Interface:
None
Phone User Interface:
None dns_cache_naptr.X.ttl
(X ranges from 1 to 12)
Permitted Values
Integer from 30 to
2147483647
Default
300
Description:
Configures the time interval (in seconds) that NAPTR record X may be cached before the record should be consulted again.
Example: dns_cache_naptr.1.ttl = 3600
Web User Interface:
None
Phone User Interface:
None dns_cache_srv.X.name
(X ranges from 1 to 12)
Domain name Blank
Description:
Configures the domain name in SRV record X.
Example: dns_cache_srv.1.name = _sip._tcp.yealink.pbx.com
Web User Interface:
None
Phone User Interface:
None dns_cache_srv.X.port
(X ranges from 1 to 12)
Integer from 0 to 65535 0
Description:
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Parameters Permitted Values
Configures the port to be used in SRV record X.
Example; dns_cache_srv.1.port = 5060
Web User Interface:
None
Phone User Interface:
None dns_cache_srv.X.priority
(X ranges from 1 to 12)
Integer from 0 to 65535
Description:
Configures the priority for the target host in SRV record X.
Lower priority is more preferred.
Web User Interface:
None
Phone User Interface:
None dns_cache_srv.X.target
(X ranges from 1 to 12)
Domain name
Default
0
Blank
Description:
Configures the domain name of the target host for an A query in SRV record X.
Example: dns_cache_srv.1.target = server1.yealink.pbx.com
Web User Interface:
None
Phone User Interface:
None dns_cache_srv.X.weight
(X ranges from 1 to 12)
Domain name 0
Description:
Configures the weight of the target host in SRV record X. When priorities are equal, weight is used to differentiate the preference.
Higher weight is more preferred.
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Parameters
Example: dns_cache_srv.1.weight = 1
Web User Interface:
None
Phone User Interface:
None dns_cache_srv.X.ttl
(X ranges from 1 to 12)
Permitted Values
Integer from 30 to
2147483647
Default
300
Description:
Configures the time interval (in seconds) that SRV record X may be cached before the record should be consulted again.
Example: dns_cache_srv.1.ttl = 3600
Web User Interface:
None
Phone User Interface:
None dns_cache_a.X.name
(X ranges from 1 to 12)
Domain name Blank
Description:
Configures the domain name in A record X.
Example: dns_cache_a.1.name = yealink.pbx.com
Web User Interface:
None
Phone User Interface:
None dns_cache_a.X.ip
(X ranges from 1 to 12)
IP address
Description:
Configures the IP address that the domain name in A record X maps to.
Example: dns_cache_a.1.ip = 192.168.1.13
Blank
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Parameters Permitted Values Default
Web User Interface:
None
Phone User Interface:
None dns_cache_a.X.ttl
(X ranges from 1 to 12)
Integer from 30 to
2147483647
300
Description:
Configures the time interval (in seconds) that A record X may be cached before the record should be consulted again.
Example: dns_cache_a.1.ttl = 3600
Web User Interface:
None
Phone User Interface:
None account.X.dns_cache_type 0, 1 or 2 1
Description:
Configures whether the IP phone uses the DNS cache for domain name resolution of the SIP server and caches the additional DNS records for account X.
0-Perform real-time DNS query rather than using DNS cache.
1-Use DNS cache, but do not cache the additional DNS records.
2-Use DNS cache and cache the additional DNS records.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.dns_cache_type = 1
Web User Interface:
None
Phone User Interface:
None account.X.static_cache_pri 0 or 1 0
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Parameters Permitted Values Default
Description:
Configures whether preferentially to use the static DNS cache for domain name resolution of the SIP server for account X.
0-Use domain name resolution from the DNS server preferentially
1-Use static DNS cache preferentially
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.static_cache_pri = 1
Web User Interface:
None
Phone User Interface:
None
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 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 provides the following capabilities for IP phones:
Capabilities Discovery -- allows LLDP-MED 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.
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Inventory Management -- provides a means to effectively manage IP phones and 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
Port ID
Mandatory TLVs
Time To Live
Description
The network address of the phone.
The MAC address of the phone.
Seconds until data unit expires.
The default value is 60s.
Optional TLVs
End of LLDPDU
System Name
Marks end of LLDPDU.
Name assigned to the IP phone.
The default value is “yealink”.
System Description
System Capabilities
Port Description
Description of the IP phone.
The default value is “yealink”.
The supported and enabled phone capabilities.
The supported capabilities are Bridge,
Telephone and Router.
The enabled capabilities are Bridge and
Telephone by default.
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)
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,
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TLV Type TLV Name
Network Policy
Description
Extended Power via MDI-PD, Inventory.
Port VLAN ID, application type, L2 priority and DSCP value.
Extended
Power-via-MDI
Inventory –
Hardware Revision
Inventory –
Firmware Revision
Inventory –
Software Revision
Inventory – Serial
Number
Power type, source, priority and value.
Hardware revision of phone.
Firmware revision of phone.
Software revision of phone.
Serial number of phone.
Inventory –
Manufacturer Name
Manufacturer name of phone.
The default value is “yealink”.
Inventory – Model
Name
Model name of phone.
Asset ID
Assertion identifier of 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
Phone User Interface
Configure LLDP feature.
Parameters: network.lldp.enable network.lldp.packet_interval
Configure LLDP feature.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
Configure LLDP feature.
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Details of Configuration Parameters:
Parameters Permitted Values Default network.lldp.enable 0 or 1 1
Description:
Enables or disables LLDP feature on the IP phone.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->LLDP->Active
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->LLDP->LLDP Status network.lldp.packet_interval Integer from 1 to 3600 60
Description:
Configures the interval (in seconds) for the IP phone to send the LLDP request.
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It works only if the value of the parameter “network.lldp.enable” is set to
1 (Enabled).
Web User Interface:
Network->Advanced->LLDP->Packet Interval (1~3600s)
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->LLDP->Packet Interval
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.
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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 the settings will take effect after reboot.
5. Click OK to reboot the phone.
To configure LLDP feature via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->LLDP->LLDP
Status.
2. Press or , or the Switch soft key to select the desired value from the LLDP
Status field.
3. Enter the desired time interval in the Packet Interval field.
4. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
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
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Administrator’s Guide for SIP-T4X IP Phones 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.
In addition to manual configuration, the IP phone also supports automatic discovery of
VLAN via LLDP or DHCP. The assignment takes effect 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.
For more information on VLAN, refer to
VLAN Feature on Yealink IP Phones
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
Procedure
VLAN can be configured using the configuration files or locally.
Configuration File <y0000000000xx>.cfg
Local Web User Interface
Configure VLAN for the Internet port and PC port manually.
Parameters: network.vlan.internet_port_enable network.vlan.internet_port_vid network.vlan.internet_port_priority network.vlan.pc_port_enable network.vlan.pc_port_vid network.vlan.pc_port_priority network.vlan.pc_port_mode
Configure DHCP VLAN discovery feature.
Parameters: network.vlan.dhcp_enable network.vlan.dhcp_option
Configure the VLAN assignment method.
Parameters: network.vlan.vlan_change.enable
Configure VLAN for the Internet
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Phone User Interface port and PC port.
Configure DHCP VLAN discovery feature.
Navigate to: http://<phoneIPAddress>/servlet?
p=network-adv&q=load
Configure VLAN for the Internet port and PC port.
Configure DHCP VLAN discovery feature.
Details of Configuration Parameters:
Parameters Permitted Values Default network.vlan.internet_port_enable 0 or 1 0
Description:
Enables or disables VLAN for the Internet (WAN) port.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->VLAN ->WAN Port->Active
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->VLAN ->WAN
Port->VLAN Status network.vlan.internet_port_vid Integer from 1 to 4094 1
Description:
Configures VLAN ID for the Internet (WAN) port.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->VLAN ->WAN Port->VID (1-4094)
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->VLAN ->WAN
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Parameters
Port->VID Number network.vlan.internet_port_priority
Permitted Values Default
Integer from 0 to 7 0
Description:
Configures VLAN priority for the Internet (WAN) port.
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.
Web User Interface:
Network->Advanced->VLAN ->WAN Port->Priority
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->VLAN ->WAN
Port->Priority network.vlan.pc_port_enable 0 or 1 0
Description:
Enables or disables VLAN for the PC (LAN) port.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->VLAN >PC Port->Active
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->VLAN ->PC Port->VLAN
Status network.vlan.pc_port_vid Integer from 1 to 4094 1
Description:
Configures VLAN ID for the PC (LAN) port.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->VLAN >PC Port->VID (1-4094)
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Configuring Advanced Features
Parameters Permitted Values Default
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->VLAN ->PC Port->VID network.vlan.pc_port_priority Integer from 0 to 7 0
Description:
Configures VLAN priority for the PC (LAN) port.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->VLAN >PC Port->Priority
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->VLAN ->PC
Port->Priority network.vlan.pc_port_mode
0 or 1 0
Description:
Configures the way the IP phone processes packets for the PC (LAN) port when
VLAN is enabled on the PC (LAN) port.
0-when packets are sent from the PC port to the Internet port, the IP phone will forward the packets directly.
1-when packets are sent from the PC port to the Internet port, and there is no VLAN tag in the packet, the IP phone will tag the packet with the tag configured for the PC port and then forward it.
Note: When packets are sent from the Internet port to the PC port, remove the packet’ tag if it is the same as the configured tag for the PC port, else forward the packets directly.
If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
None
Phone User Interface:
None network.vlan.dhcp_enable 0 or 1 1
Description:
Enables or disables DHCP VLAN discovery feature on the IP phone.
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Parameters Permitted Values Default
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->VLAN >DHCP VLAN->Active
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->VLAN ->DHCP VLAN
->DHCP VLAN network.vlan.dhcp_option Integer from 128 to 254 132
Description:
Configures the DHCP option from which the IP phone will obtain the VLAN settings.
You can configure at most five DHCP options and separate them by commas.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->VLAN->DHCP VLAN->Option(1-255)
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->VLAN ->DHCP VLAN ->
Option network.vlan.vlan_change.enable 0 or 1 0
Description:
Enables or disables the IP phone to obtain IP address with lower preference of
VLAN assignment method or disable VLAN feature when the IP phone cannot obtain IP address with the current VLAN assignment method.
0-Disabled
1-Enabled
Web User Interface:
None
Phone User Interface:
None
To configure VLAN for Internet port via web user interface:
1. Click on Network->Advanced.
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Configuring Advanced Features
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.
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 reboot to make the settings effective.
6. Click OK to reboot the 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 settings will take effect after reboot.
6. Click OK to reboot the phone.
To configure the DHCP VLAN discovery via web user interface:
1. Click on Network->Advanced.
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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 (1-255) field.
The default option is 132.
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4. Click Confirm to accept the change.
A dialog box pops up to prompt that settings will take effect after reboot.
5. Click OK to reboot the phone.
To configure VLAN for Internet port (or PC port) via phone user interface:
1. Press Menu->Advanced (default 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 field.
4. Enter the priority value (0-7) in the Priority field.
5. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
To configure DHCP VLAN discovery via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->VLAN->DHCP
VLAN.
2. Press or , or the Switch soft key to select the desired value from the DHCP
VLAN field.
3. Enter the desired option in the Option field.
4. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
Configuring Advanced Features
VPN (Virtual Private Network) is a secured private network connection built on top of public telecommunication infrastructure, such as the Internet. VPN has become more prevalent due to the benefits of scalability, reliability, convenience and security. VPN 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 networking 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 the VPN feature. To prevent disclosure of private information, tunnel endpoints must authenticate each other before secure VPN tunnel is established. After the 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
VPN-related 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
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
Procedure
VPN can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the OpenVPN feature and upload a TAR file to the IP phone.
Parameters: network.vpn_enable openvpn.url
Configure VPN feature and upload a TAR package to the IP phone.
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Phone User Interface
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
Configure VPN feature.
Details of Configuration Parameters:
Parameters Permitted Values Default network.vpn_enable 0 or 1 0
Description:
Enables or disables OpenVPN feature on the IP phone.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->VPN ->Active
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->VPN->VPN Active openvpn.url URL within 511 characters Blank
Description:
Configures the access URL of the *.tar file for OpenVPN.
Example: openvpn.url = http://192.168.10.25/OpenVPN.tar
Web User Interface:
Network->Advanced->VPN->Upload VPN Config
Phone User Interface:
None
To upload the tar file to the phone and configure VPN via web user interface:
1. Click on Network->Advanced.
2. Click Browse to locate the TAR package from the local system.
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3. Click Upload to upload the TAR file.
Configuring Advanced Features
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 reboot.
6. Click OK to reboot the phone.
To configure VPN via phone user interface after uploading the tar file:
1. Press Menu->Advanced (default 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 LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
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
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Administrator’s Guide for SIP-T4X IP Phones 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 the 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 to be delayed or dropped due to interference from other lower priority traffic. VoIP can guarantee high-quality QoS only if the voice and the SIP packets are given priority over other kinds of network traffic. IP phones support the DiffServ model of QoS.
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Configuring Advanced Features
Voice QoS
In order to make VoIP transmissions intelligible to receivers, voice packets should not be dropped, excessively delayed, 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
DSCPs for voice packets and SIP packets 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.
Parameters: network.qos.rtptos network.qos.signaltos
Configure the DSCPs for voice packets and SIP packets.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default network.qos.rtptos Integer from 0 to 63 46
Description:
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.
Web User Interface:
Network->Advanced->Voice QoS (0~63)
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Parameters
Phone User Interface:
None network.qos.signaltos
Permitted Values Default
Integer from 0 to 63 26
Description:
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.
Web User Interface:
Network->Advanced->SIP QoS (0~63)
Phone User Interface:
None
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.
3. Enter the desired value in the SIP Qos (0~63) field.
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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 reboot.
5. Click OK to reboot the phone.
Configuring Advanced Features
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. The NAT feature 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 assistance from a third-party network server (STUN server) usually located on public
Internet. The IP phone can be configured to act as a STUN client, sending 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.
Parameters: account.X.nat.nat_traversal account.X.nat.stun_server account.X.nat.stun_port
Configure NAT traversal and
STUN server on the IP phone.
Navigate to: http://<phoneIPAddress>/servl et?p=account-register&q=load
&acc=0
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Details of Configuration Parameters:
Parameters Permitted Values account.X.nat.nat_traversal 0 or 1
Description:
Enables or disables the NAT traversal for account X.
0-Disabled
1-Enabled
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Register ->NAT
Phone User Interface:
None account.X.nat.stun_server IP address or domain name
Default
0
Blank
Description:
Configures the IP address or the domain name of the STUN server for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.nat.stun_server = 218.107.220.201
Web User Interface:
Account->Register ->STUN Server
Phone User Interface:
None account.X.nat.stun_port Integer from 1024 to 65000 3478
Description:
Configures the port of the STUN server for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
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Parameters Permitted Values
Example: account.1.nat.stun_port = 3478
Web User Interface:
Account->Register ->STUN Server ->Port
Phone User Interface:
None
To configure the NAT traversal and STUN server via web user interface:
1. Click on Account.
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 in the STUN Server field.
Default
5. Click Confirm to accept the change.
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 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
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IP phones support protocols EAP-MD5, EAP-TLS, EAP-PEAP/MSCHAPv2,
EAP-TTLS/EAP-MSCHAPv2, EAP-PEAP/GTC and EAP-TTLS/EAP-GTC for 802.1X authentication.
For more information on 802.1X authentication, refer to
Yealink 802.1X Authentication
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
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.
Parameters: network.802_1x.mode network.802_1x.identity network.802_1x.md5_password network.802_1x.root_cert_url network.802_1x.client_cert_url.
Configure the 802.1X authentication.
Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load
Configure the 802.1X authentication.
Details of Configuration Parameters:
Parameters Permitted Values Default
0 network.802_1x.mode
Description:
Configures the 802.1x authentication method.
0-Disabled
1-EAP-MD5
2-EAP-TLS
0, 1, 2, 3, 4, 5 or 6
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Configuring Advanced Features
Parameters Permitted Values Default
3-EAP-PEAP/MSCHAPv2
4-EAP-TTLS/EAP-MSCHAPv2
5-EAP-PEAP/GTC
6-EAP-TTLS/EAP-GTC
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->802.1x->802.1x Mode
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->802.1x >802.1x Mode network.802_1x.identity String within 32 characters Blank
Description:
Configures the user name for 802.1x authentication.
Example: network.802_1x.identity = yealink
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Advanced->802.1x->Identity
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->802.1x >802.1x Mode
->Identity network.802_1x.md5_password String within 32 characters Blank
Description:
Configures the password for 802.1x authentication.
Example: network.802_1x.md5_password = admin123
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is required for all 802.1x authentication methods except EAP-TLS.
Web User Interface:
Network->Advanced->802.1x->MD5 Password
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->802.1x ->802.1x Mode
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Parameters
(EAP-MD5)->MD5 assword network.802_1x.root_cert_url
Permitted Values Default
URL within 511 characters Blank
Description:
Configures the access URL of the CA certificate when the 802.1x authentication method is configured as EAP-TLS, EAP-PEAP/MSCHAPv2, EAP-TTLS/EAP-MSCHAPV2,
EAP-PEAP/GTC or EAP-TTLS/EAP-GTC .
Example : network.802_1x.root_cert_url = http://192.168.1.10/ca.pem
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is only applicable to EAP-TLS, EAP-PEAP/MSCHAPv2,
EAP-TTLS/EAP-MSCHAPV2, EAP-PEAP/GTC and EAP-TTLS/EAP-GTC protocols. The format of the certificate must be *.pem, *.crt, *.cer or *.der.
Web User Interface:
Network->Advanced->802.1x->CA Certificates
Phone User Interface:
None network.802_1x.client_cert_url URL within 511 characters Blank
Description:
Configures the access URL of the device certificate when the 802.1x authentication method is configured as EAP-TLS.
Example: network.802_1x.client_cert_url = http://192.168.1.10/client.pem
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.
Web User Interface:
Network->Advanced->802.1x->Device Certificates
Phone User Interface:
None
To configure the 802.1X 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 Mode
802.1x.
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Configuring Advanced Features 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.
3) In the CA Certificates field, click Browse to locate the desired CA certificate (*.pem,*.crt, *.cer or *.der) from your local system.
4) In the Device Certificates field, click Browse to locate the desired client certificate (*.pem or *.cer) from your local system.
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5) Click Upload to upload the certificates. c) If you select EAP-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 locate the desired certificate
(*.pem,*.crt, *.cer or *.der) from your local system.
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4) Click Upload to upload the certificate.
Configuring Advanced Features 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 locate the desired certificate
(*.pem,*.crt, *.cer or *.der) from your local system.
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4) Click Upload to upload the certificate. e) If you select EAP-PEAP/GTC:
1) Enter the user name for authentication in the Identity field.
2) Enter the password for authentication in the MD5 Password field.
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Configuring Advanced Features
3) In the CA Certificates field, click Browse to select the desired CA certificate
(*.pem, *.crt, *.cer or *.der) from your local system.
4) Click Upload to upload the certificate. f) If you select EAP-TTLS/EAP-GTC:
1) Enter the user name for authentication in the Identity field.
2) Enter the password for authentication in the MD5 Password field.
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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 the settings will take effect after reboot.
4. Click OK to reboot the phone.
To configure the 802.1X via phone user interface after:
1. Press Menu->Advanced (default password: admin) ->Network->802.1x.
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.
Configuring Advanced Features c) If you select EAP-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.
2) Enter the password for authentication in the MD5 Password field. e) If you select EAP-PEAP/GTC:
1) Enter the user name for authentication in the Identity field.
2) Enter the password for authentication in the MD5 Password field. f) If you select EAP-TTLS/EAP-GTC:
1) Enter the user name for authentication in the Identity field.
2) Enter the password for authentication in the MD5 Password field.
3. Click Save to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
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
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.
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GetParameterValues
GetParameterNames
GetParameterAttributes
SetParameterAttributes
Reboot
RPC Method
Download
Upload
ScheduleInform
FactoryReset
TransferComplete
AddObject
DeleteObject
Description
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.
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.
For more information on TR-069, refer to
Yealink TR-069 Technote
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
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Configuring Advanced Features
Procedure
TR-069 can be configured using the configuration files or locally.
Configurati on File
Local
<y0000000000xx>.cfg
Configure theTR-069 feature.
Parameters: managementserver.enable managementserver.username managementserver.password managementserver.url managementserver.connection_request_use
rname managementserver.connection_request_pas
sword managementserver.periodic_inform_enable managementserver.periodic_inform_interval
Web User Interface
Configure the TR-069 feature.
Navigate to: http://<phoneIPAddress>/servlet?p=settings
-preference&q=load
Details of Configuration Parameters:
Parameters
Permitted
Values
Default managementserver.enable 0 or 1 0
Description:
Enables or disables TR-069 feature.
0-Disabled
1-Enabled
Web User Interface:
Settings->TR069->Enable TR069
Phone User Interface:
None managementserver.username
String within
128 characters
Blank
Description:
Configures the user name for the IP phone to authenticate with the ACS (Auto
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Parameters
Permitted
Values
Default
Configuration Servers). This string is set to the empty string if no authentication is required.
Example: managementserver.username = user1
Web User Interface:
Settings->TR069->ACS Username
Phone User Interface:
None managementserver.password
String within
64 characters
Blank
Description:
Configures the password for the IP phone to authenticate with the ACS (Auto
Configuration Servers). This string is set to the empty string if no authentication is required.
Example: managementserver.password = pwd123
Web User Interface:
Settings->TR069->ACS Password
Phone User Interface:
None managementserver.url
URL within 511 characters
Blank
Description:
Configures the access URL of the ACS (Auto Configuration Servers).
Example: managementserver.url = http://192.168.1.20/acs/
Web User Interface:
Settings->TR069->ACS URL
Phone User Interface:
None managementserver.connection_request_username
String within
128 characters
Blank
Description:
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Parameters
Permitted
Values
Default
Configures the user name for the IP phone to authenticate the incoming connection requests.
Example: managementserver.connection_request_username = accuser
Web User Interface:
Settings->TR069->Connection Request Username
Phone User Interface:
None managementserver.connection_request_password
String within
64 characters
Blank
Description:
Configures the password for the IP phone to authenticate the incoming connection requests.
Example: managementserver.connection_request_password = acspwd
Web User Interface:
Settings->TR069->Connection Request Password
Phone User Interface:
None managementserver.periodic_inform_enable 0 or 1 1
Description:
Enables or disables the IP phone to periodically report its configuration information to the ACS (Auto Configuration Servers).
0-Disabled
1-Enabled
Web User Interface:
Settings->TR069->Enable Periodic Inform
Phone User Interface:
None managementserver.periodic_inform_interval
Integer from 5 to 4294967295
60
Description:
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Parameters
Permitted
Values
Default
Configures the interval (in seconds) for the IP phone to report its configuration to the
ACS (Auto Configuration Servers).
Web User Interface:
Settings->TR069->Periodic Inform Interval (seconds)
Phone User Interface:
None
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.
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
Configuring Advanced Features 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.
Stateful DHCPv6: The Dynamic Host Configuration Protocol for IPv6 (DHCPv6) has been standardized by the IETF through RFC3315. DHCPv6 enables DHCP servers to pass configuration parameters such as IPv6 network addresses to IPv6 nodes. It offers the capability of automatic allocation of reusable network addresses and additional configuration flexibility. This protocol is a stateful counterpart to “IPv6
Stateless Address Autoconfiguration”, and can be used separately or in addition to the stateless autoconfiguration to obtain configuration parameters.
Note
SIP-T42G and SIP-T41P IP phones do not support the stateful DHCPv6 address assignment method.
If the IP phone enables the SLAAC and DHCPv6 features simultaneously, the IP phone will obtain the IP address via SLAAC and obtain other network parameters via DHCPv6.
Procedure
IPv6 can be configured using the configuration files or locally.
Configuration File <MAC>.cfg
Configure the IPv6 address assignment method.
Parameters: network.ip_address_mode network.ipv6_internet_port.type network.ipv6_internet_port.ip network.ipv6_prefix network.ipv6_internet_port.gateway
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Local
<y0000000000xx>.cf
g
Web User Interface
Phone User Interface network.ipv6_icmp_v6.enable
Configure the IPv6 static DNS address. network.ipv6_primary_dns network.ipv6_secondary_dns
Configure the IPv6 static DNS.
Parameter: network.ipv6_static_dns_enable
Configure the IPv6 address assignment method.
Configure the IPv6 static DNS.
Navigate to: http://<phoneIPAddress>/servlet?p
=network&q=load
Configure the IPv6 address assignment method.
Configure the IPv6 static DNS.
Details of Configuration Parameters:
Parameters Permitted Values Default network.ip_address_mode 0, 1 or 2 0
Description:
Configures the IP address mode.
0-IPv4
1-IPv6
2-IPv4&IPv6
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->Internet Port->Mode (IPv4/IPv6)
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN Port->IP Address
Mode network.ipv6_internet_port.type 0 or 1 0
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Parameters Permitted Values Default
Description:
Configures the Internet (WAN) port type for IPv6 when the IP address mode is configured as IPv6 or IPv4&IPv6.
0-DHCP
1-Static IP Address
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv6 Config
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN Port->IPv6->Type network.ipv6_static_dns_enable 0 or 1 0
Description:
Enables or disables the IP phone to use manually configured static IPv6 DNS when
Internet (WAN) port type for IPv6 is configured as DHCP.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv6 Config->IPv6 Static DNS
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN Port->IPv6->Static
DNS network.ipv6_internet_port.ip IPv6 address Blank
Description:
Configures the IPv6 address when the IP address mode is configured as IPv6 or
IPv4&IPv6, and the Internet (WAN) port type for IPv6 is configured as Static IP
Address.
Example: network.ipv6_internet_port.ip = 2005:1:1:1::12
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Parameters Permitted Values
Web User Interface:
Network->Basic->IPv6 Config->Static IP Address->IP Address
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv6->Type(Static IP) ->IP Address network.ipv6_prefix Integer from 0 to 128
Default
64
Description:
Configures the IPv6 prefix when the IP address mode is configured as IPv6 or
IPv4&IPv6, and the Internet (WAN) port type for IPv6 is configured as Static IP
Address.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv6 Config->Static IP Address->IPv6 Prefix (0~128)
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv6->Type(Static IP) ->IPv6 IP Prefix network.ipv6_internet_port.gateway IPv6 address Blank
Description:
Configures the IPv6 default gateway when the IP address mode is configured as IPv6 or
IPv4&IPv6, and the Internet (WAN) port type for IPv6 is configured as Static IP Address.
Example: network.ipv6_internet_port.gateway = 2005:1:1:1::1
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv6 Config->Static IP Address->Gateway
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv6->Type(Static IP) ->Gateway network.ipv6_primary_dns IPv6 address Blank
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Configuring Advanced Features
Parameters Permitted Values Default
Description:
Configures the primary IPv6 DNS server when the IP address mode is configured as
IPv6 or IPv4&IPv6, and the Internet (WAN) port type for IPv6 is configured as Static IP
Address, or and the Internet (WAN) port type for IPv6 is configured as DHCP and
Staic DNS is configured as Enabled.
Example: network.ipv6_primary_dns = 2005:1:1:1::89
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv6 Config->IPv6 Static DNS->Primary DNS
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv6->Type(Static IP) ->Primary DNS
Or Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv6->Type(DHCP) ->Staic DNS (Enabled) ->Primary DNS network.ipv6_secondary_dns IPv6 address Blank
Description:
Configures the secondary IPv6 DNS server when the IP address mode is configured as IPv6 or IPv4&IPv6, and the Internet (WAN) port type for IPv6 is configured as Static
IP Address, or and the Internet (WAN) port type for IPv6 is configured as DHCP and
Staic DNS is configured as Enabled.
Example: network.ipv6_secondary_dns = 2005:1:1:1::87
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Network->Basic->IPv6 Config->IPv6 Static DNS->Secondary DNS
Phone User Interface:
Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv6->Type(Static IP) ->Secondary DNS
Or Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv6->Type(DHCP) ->Staic DNS (Enabled) ->Secondary DNS network.ipv6_icmp_v6.enable 0 or 1 1
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Parameters Permitted Values Default
Description:
Enables or disables the IP phone to obtain IPv6 network settings via SLAAC
(Stateless Address Autoconfiguration) method.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is only applicable to SIP-T48G/T46G IP phones.
Web User Interface:
Network->Advanced->ICMPv6 Status->Active
Phone User Interface:
None
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, mark the DHCP or the Static IP Address radio box.
-
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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Configuring Advanced Features
-
(Optional.) If you mark the DHCP radio box, you can configure the static DNS address in the corresponding fields.
4. Click Confirm to accept the change.
A dialog box pops up to prompt that the settings will take effect after reboot.
5. Click OK to reboot the phone.
To configure SLAAC feature via web user interface (not applicable to SIP-T42G/T41P):
1. Click on Network->Advanced.
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2. In the ICMPv6 Status block, select the desired value from the pull-down list of
Active.
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3. Click Confirm to accept the change.
To configure IPv6 address via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->WAN Port.
2. Press or to select the desired address mode from the IP Address 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 IP, configure the IPv6 address and other configuration parameters in the corresponding fields.
5. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
To configure static IPv6 DNS when DHCPv6 is used via phone user interface:
1. Press Menu->Advanced (default password: admin) ->Network->WAN
Port->IPv6->Static DNS.
2. Press or , or the Switch soft key to select Enabled from the Static DNS field.
3. Enter the desired values in the Primary DNS and Secondary DNS fields respectively.
4. Press the Save soft key to accept the change.
The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel.
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.
Parameter: features.headset_prior
Configure headset prior.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default
0 features.headset_prior
Description:
Enables or disables headset prior feature.
0-Disabled
1-Enabled
0 or 1
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Parameter Permitted Values Default
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.
Web User Interface:
Features->General Information->Headset Prior
Phone User Interface:
None
To configure headset prior via web user interface:
1. Click on Features->General Information.
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 IP 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.
Configuring Audio Features
Procedure
Dual headset can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure dual headset.
Parameter: features.headset_training
Configure dual headset.
Navigate to: http://<phoneIPAddress>/servlet
?p=features-general&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default features.headset_training 0 or 1 0
Description:
Enables or disables dual headset feature.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), users can use two headsets on one phone. When the IP phone joins in a call, 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.
Web User Interface:
Features->General Information->Dual-Headset
Phone User Interface:
None
To configure dual headset 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 Dual-Headset.
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3. Click Confirm to accept the change.
Voice quality monitoring feature allows the IP phones to generate various quality metrics for listening quality and conversational quality. These metrics can be sent between the phones in RTCP-XR packets. These metrics can also be sent in SIP PUBLISH messages to a central voice quality report collector.
Two methods for voice quality monitoring are supported by Yealink IP phones:
RTCP-XR
VQ-RTCPXR
The RTCP-XR method, complaint with RFC 3611-RTP Control Extended Reports (RTCP-XR) , provides the metrics contained in RTCP-XR packets for monitoring and diagnosing the call quality and performance. These metrics include network packet loss, delay metrics, analog metrics and voice quality metrics.
Configuring Audio Features
Procedure
RTCP-XR can be configured using the configuration files.
Configuration
File
<y0000000000xx>.cfg
Configure the generation of RTCP-XR packets.
Parameter: phone_setting.rtcp_xr_report.enable
Details of Configuration Parameters:
Parameters Permitted Values Default phone_setting.rtcp_xr_report.enable 0 or 1 0
Description:
Enables or disables the IP phone to periodically (every 5 seconds) send RTCP-XR packets to another participating party during a call for call quality monitoring and diagnosing.
Web User Interface:
None
Phone User Interface:
None
The VQ-RTCPXR method, complaint with RFC 6035 , sends the service quality metric reports contained in SIP PUBLISH messages to the central report collector. Three types of quality reports can be enabled:
Session: Generated at the end of a call.
Interval: Generated during a call at a configurable period.
Alert: Generated when the call quality degrades below a configurable threshold.
A wide range of performance metrics are generated in the following two ways:
Based on current values, such as jitter, jitter buffer max and round trip delay.
Computed using other metrics as input, such as listening Mean Opinion Score
(MOS-LQ) and conversational Mean Opinion Score (MOS-CQ).
To operate with central report collector, IP phones must be configured to forward their voice quality reports to the specified report collector. You can specify the report collector on a per-line basis.
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Users can check the voice quality data of the last call via web user interface or phone user interface. Users can also specify the options of the RTP status to be displayed on the phone user interface. Options of the RTP status to be displayed on the web user interface cannot be specified.
Note
When using voice quality monitoring feature, some problems will occur:
1. GapDuration always equals to 0 while no burst duration.
2. JitterBufferAdaptive always equals to 2 (non-adaptive/fixed), even if it's configured adaptive.
3. MOSLQ/MOSCQ may be lower or higher than what VQMon calculates sometimes
( error of [1, +0.5]).
The problems will be fixed in firmware version 80.
Procedure
RTCP-XR can be configured using the configuration files or locally.
Configuration
File
<y0000000000xx>
.cfg
Configure the generation of session packets.
Parameter: phone_setting.vq_rtcpxr.session_report.enable
Configure the generation of interval packets.
Parameters: phone_setting.vq_rtcpxr.interval_report.enabl
e phone_setting.vq_rtcpxr_interval_period
Configure the generation of alert packets.
Parameters: phone_setting.vq_rtcpxr_moslq_threshold_war
ning phone_setting.vq_rtcpxr_moslq_threshold_criti
cal phone_setting.vq_rtcpxr_delay_threshold_war
ning phone_setting.vq_rtcpxr_delay_threshold_criti
cal
Configure the phone to display RTP status showing the voice quality report of the last call on the web user interface.
Parameter: phone_setting.vq_rtcpxr.states_show_on_web.
enable
Configure the phone to display RTP status
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Local
<MAC>.cfg
Web User
Configuring Audio Features showing the voice quality report of the last call or the current call on the phone user interface.
Parameter: phone_setting.vq_rtcpxr.states_show_on_gui.e
nable
Configure the options of the RTP status displayed on the phone user interface.
Parameters: phone_setting.vq_rtcpxr_display_start_time.en
able phone_setting.vq_rtcpxr_display_stop_time.en
able phone_setting.vq_rtcpxr_display_local_call_id.
enable phone_setting.vq_rtcpxr_display_remote_call_
id.enable phone_setting.vq_rtcpxr_display_local_codec.
enable phone_setting.vq_rtcpxr_display_remote_cod
ec.enable phone_setting.vq_rtcpxr_display_jitter.enable phone_setting.vq_rtcpxr_display_jitter_buffer_
max.enable phone_setting.vq_rtcpxr_display_packets_lost.
enable phone_setting.vq_rtcpxr_display_symm_onew
ay_delay.enable phone_setting.vq_rtcpxr_display_round_trip_d
elay.enable phone_setting.vq_rtcpxr_display_moslq.enabl
e phone_setting.vq_rtcpxr_display_moscq.enabl
e
Configure the central report collector.
Parameters: account.X.vq_rtcpxr.collector_name account.X.vq_rtcpxr.collector_server_host account.X.vq_rtcpxr.collector_server_port
Configure VQ-RTCPXR.
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Interface Configure the phone to display RTP status showing the voice quality report of the last call on the web user interface.
Configure the phone to display RTP status showing the voice quality report of the last call or the current call on the phone user interface.
Configure the options of the RTP status displayed on the phone user interface.
Navigate to: http://<phoneIPAddress>/servlet?p=settings-v oicemonitoring&q=load
Configure the central report collector.
Navigate to: http://<phoneIPAddress>/servlet?p=accountadv&q=load&acc=0
Details of Configuration Parameters:
Parameters
Permitted
Values
Default phone_setting.vq_rtcpxr.session_report.enable 0 or 1 0
Description:
Enables or disables the IP phone to send a session quality report to the central report collector at the end of each call.
0-Disabled
1-Enabled
Web User Interface:
Settings->Voice Monitoring->VQ RTCP-XR Session Report
Phone User Interface:
None phone_setting.vq_rtcpxr.interval_report.enable 0 or 1 0
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Configuring Audio Features
Parameters
Permitted
Values
Default
Description:
Enables or disables the IP phone to send an interval quality report to the central report collector periodically throughout a call.
0-Disabled
1-Enabled
Web User Interface:
Settings->Voice Monitoring->VQ RTCP-XR Interval Report
Phone User Interface:
None phone_setting.vq_rtcpxr_interval_period
Integer from
5 to 20
20
Description:
Configures the interval (in seconds) for the IP phone to send an interval quality report to the central report collector periodically throughout a call.
Web User Interface:
Settings->Voice Monitoring->Period for Interval Report
Phone User Interface:
None phone_setting.vq_rtcpxr_moslq_threshold_warning 15 to 40 Blank
Description:
Configures the threshold value of listening MOS score (MOS-LQ) multiplied by 10.
The threshold value of MOS-LQ causes the phone to send a warning alert quality report to the central report collector.
For example, a configured value of 35 corresponds to the MOS score 3.5. When the
MOS-LQ value computed by the phone is less than or equal to 3.5, the phone will send a warning alert quality report to the central report collector. When the MOS-LQ value computed by the phone is greater than 3.5, the phone will not send a warning alert quality report to the central report collector.
If it is set to blank, warning alerts are not generated due to MOS-LQ.
Web User Interface:
Settings->Voice Monitoring->Warning threshold for Moslq
Phone User Interface:
None
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Parameters
Permitted
Values
Default
15 to 40 Blank phone_setting.vq_rtcpxr_moslq_threshold_critical
Description:
Configures the desired threshold value of listening MOS score (MOS-LQ) multiplied by 10. The threshold value of MOS-LQ causes the phone to send a critical alert quality report to the central report collector.
For example, a configured value of 28 corresponds to the MOS score 2.8. When the
MOS-LQ value computed by the phone is less than or equal to 2.8, the phone will send a critical alert quality report to the central report collector. When the MOS-LQ value computed by the phone is greater than 2.8, the phone will not send a critical alert quality report to the central report collector.
If it is set to blank, critical alerts are not generated due to MOS-LQ.
Web User Interface:
Settings->Voice Monitoring->Critical threshold for Moslq
Phone User Interface:
None phone_setting.vq_rtcpxr_delay_threshold_warning 10 to 2000 Blank
Description:
Configures the threshold value of one way delay (in ms) that causes the phone to send a warning alert quality report to the central report collector.
For example, If it is set to 500, when the value of one way delay computed by the phone is greater than or equal to 500, the phone will send a warning alert quality report to the central report collector; when the value of one way delay computed by the phone is less than 500, the phone will not send a warning alert quality report to the central report collector.
If it is set to blank, warning alerts are not generated due to one way delay. One-way delay includes both network delay and end system delay.
Web User Interface:
Settings->Voice Monitoring->Warning threshold for Delay
Phone User Interface:
None phone_setting.vq_rtcpxr_delay_threshold_critical 10 to 2000 Blank
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Parameters
Permitted
Values
Default
Description:
Configures the threshold value of one way delay (in ms) that causes phone to send a critical alert quality report to the central report collector.
For example, If it is set to 500, when the value of one way delay computed by the phone is greater than or equal to 500, the phone will send a critical alert quality report to the central report collector; when the value of one way delay computed by the phone is less than 500, the phone will not send a critical alert quality report to the central report collector.
If it is set to blank, critical alerts are not generated due to one way delay. One-way delay includes both network delay and end system delay.
Web User Interface:
Settings->Voice Monitoring->Critical threshold for Delay
Phone User Interface:
None phone_setting.vq_rtcpxr.states_show_on_web.enable 0 or 1 0
Description:
Enables or disables the voice quality data of the last call to be displayed on web interface at path Status->RTP Status.
0-Disabled
1-Enabled
Web User Interface:
Settings->Voice Monitoring->Display Report options on Web
Phone User Interface:
None phone_setting.vq_rtcpxr.states_show_on_gui.enable 0 or 1 0
Description:
Enables or disables the voice quality data of the last call or current call to be displayed on the LCD screen. You can view the voice quality data of the last call by pressing Menu->Status->RTP Status. You can view the voice quality data of the current call by pressing RTP Status soft key during a call.
0-Disabled
1-Enabled
Web User Interface:
Settings->Voice Monitoring->Display Report options on phone
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Parameters
Phone User Interface:
None phone_setting.vq_rtcpxr_display_start_time.enable
Permitted
Values
0 or 1
Description:
Enables or disables the phone to display Start Time on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->Start Time
Phone User Interface:
None phone_setting.vq_rtcpxr_display_stop_time.enable 0 or 1
Default
1
1
Description:
Enables or disables the phone to display Current Time or Stop Time on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->Current Time
Phone User Interface:
None phone_setting.vq_rtcpxr_display_local_call_id.enable 0 or 1 1
Description:
Enables or disables the phone to display Local User on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
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Parameters
Permitted
Values
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->Local User
Phone User Interface:
None
Default phone_setting.vq_rtcpxr_display_remote_call_id.enable 0 or 1 1
Description:
Enables or disables the phone to display Remote User on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->Remote User
Phone User Interface:
None phone_setting.vq_rtcpxr_display_local_codec.enable 0 or 1
Description:
Enables or disables the phone to display Local Codec on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->Local Codec
Phone User Interface:
None phone_setting.vq_rtcpxr_display_remote_codec.enable 0 or 1
1
1
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Parameters
Permitted
Values
Default
Description:
Enables or disables the phone to display Remote Codec on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->Remote Codec
Phone User Interface:
None phone_setting.vq_rtcpxr_display_jitter.enable 0 or 1 1
Description:
Enables or disables the phone to display Jitter on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->Jitter
Phone User Interface:
None phone_setting.vq_rtcpxr_display_jitter_buffer_max.enable 0 or 1 1
Description:
Enables or disables the phone to display JitteBufferMax on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->JitteBufferMax
Phone User Interface:
None
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Parameters phone_setting.vq_rtcpxr_display_packets_lost.enable
Permitted
Values
Default
0 or 1 1
Description:
Enables or disables the phone to display Packet lost on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->Packet lost
Phone User Interface:
None phone_setting.vq_rtcpxr_display_symm_oneway_delay.ena
ble
0 or 1 0
Description:
Enables or disables the phone to display SymmOneWayDelay on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->SymmOneWayDelay
Phone User Interface:
None phone_setting.vq_rtcpxr_display_round_trip_delay.enable 0 or 1 0
Description:
Enables or disables the phone to display RoundTripDelay on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->RoundTripDelay
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Parameters
Phone User Interface:
None phone_setting.vq_rtcpxr_display_moslq.enable
Permitted
Values
0 or 1
Default
1
Description:
Enables or disables the phone to display MOS-LQ on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->MOS-LQ
Phone User Interface:
None phone_setting.vq_rtcpxr_display_moscq.enable 0 or 1 1
Description:
Enables or disables the phone to display MOS-CQ on the LCD screen.
0-Disabled
1-Enabled
Note: It works only if the value of the parameter
“phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.
Web User Interface:
Settings->Voice Monitoring->Report options on phone UI->MOS-CQ
Phone User Interface:
None account.X.vq_rtcpxr.collector_name
String within 32 characters
Blank
Description:
Configures the host name of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
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Configuring Audio Features
Parameters
Permitted
Values
Default
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->VQ RTCP-XR Collector name
Phone User Interface:
None account.X.vq_rtcpxr.collector_server_host
IPv4
Address
Blank
Description:
Configures the IP address of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->VQ RTCP-XR Collector address
Phone User Interface:
None account.X.vq_rtcpxr.collector_server_port
Integer from 1 to
65535
5060
Description:
Configures the port of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->VQ RTCP-XR Collector port
Phone User Interface:
None
To configure session report for VQ-RTCPXR via web user interface:
1. Click on Settings->Voice Monitoring.
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2. Select the desired value from the pull-down list of VQ RTCP-XR Session Report.
3. Click Confirm to accept the change.
To configure interval report for VQ RTCP-XR via web user interface:
1. Click on Settings->Voice Monitoring.
2. Select the desired value from the pull-down list of VQ RTCP-XR Interval Report.
3. Enter the desired value in the Period for Interval Report field.
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4. Click Confirm to accept the change.
Configuring Audio Features
To configure alert report for VQ RTCP-XR via web user interface:
1. Click on Settings->Voice Monitoring.
2. Enter the desired value in the Warning threshold for Moslq field.
3. Enter the desired value in the Critical threshold for Moslq field.
4.
Enter the desired value in the Warning threshold for Delay field.
5. Enter the desired value in the Critical threshold for Delay field.
6. Click Confirm to accept the change.
To configure RTP status displayed on the web page via web user interface:
1. Click on Settings->Voice Monitoring.
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2. Select the desired value from the pull-down list of Display Report options on Web.
3. Click Confirm to accept the change.
The RTP status will appear on the web user interface at the path: Status.
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To configure RTP status displayed on the LCD screen via web user interface:
1. Click on Settings->Voice Monitoring.
Configuring Audio Features
2. Select the desired value from the pull-down list of Display Report options on phone.
3. Click Confirm to accept the change.
The RTP status will appear on the phone user interface at the path:
Menu->Status->More….
To configure the options of the RTP status displayed on the LCD screen via web user interface:
1. Click on Settings->Voice Monitoring.
2. In the Report options on phone UI block, select the desired list from the Disabled column and then click .
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The selected list appears in the Enabled column.
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3. Repeat step 2 to add more items to the Enabled column.
4. To remove an item from the Enabled column, select the desired item and then click .
5. To adjust the display order of enabled items, select the desired item and then click or .
The LCD screen will display the item(s) in the adjusted order.
6. Click Confirm to accept the change.
To configure the central report collector via web user interface:
1. Click on Account->Advanced.
2.
Enter the host name of the central report collector in the VQ RTCP-XR Collector name field.
3.
Enter the IP address of the central report collector in the VQ RTCP-XR Collector address field.
Configuring Audio Features
4. Enter the port of the central report collector in the VQ RTCP-XR Collector port field.
5. 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 following table lists the audio codecs supported by SIP-T4X IP phones:
Supported Audio Codecs
G722, PCMA, PCMU, G729, G723_53,
G723_63, G726-16, G726-24, G726-32,
G726-40, iLBC
Default Audio Codecs
G722, PCMA, PCMU, G729
The following table summarizes the supported audio codecs on IP phones:
Sample Packetization Codec
G722
PCMA
PCMU
G729
G726-16
Algorithm Reference
G.722 RFC 3551
G.711 RFC 3551
RFC 3551
G.726
RFC 3551
RFC 3551
Bit Rate
64 Kbps
64 Kbps
64 Kbps
8 Kbps
16 Kbps
8 Ksps
8 Ksps
8 Ksps
8 Ksps
20ms
20ms
20ms
20ms
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Codec
G726-24
G726-32
G726-40
G723_53/
G723_63 iLBC
Algorithm Reference
G.726 RFC 3551
G.726
G.726
RFC 3551
RFC 3551
G.723.1 iLBC
RFC 3951
RFC 3952
Bit Rate
24 Kbps
32 Kbps
40 Kbps
5.3kbps
6.3kbps
13.33 Kbps
15.2 Kbps
Sample Packetization
8 Ksps
8 Ksps
8 Ksps
8 Ksps
20ms
20ms
30ms
20ms
30ms
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.
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.
The corresponding attributes of the codec are listed as follows:
Codec
G722
PCMU
PCMA
G729
G723_53
G723_63
G726-16
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
Priority
1
2
3
4
0
0
0
RTPmap
9
0
8
18
4
4
103
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Configuring Audio Features
Codec
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
Priority
0
0
0
0
RTPmap
104
102
105
106
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.
Parameters: account.X.codec.Y.enable account.X.codec.Y.payload_type
Configure the priority and rtpmap for the enabled codec.
Parameters: account.X.codec.Y.priority account.X.codec.Y.rtpmap
Configure the ptime.
Parameter: account.X.ptime
Configure the codecs and adjust the priority of the enabled codecs on a per-line basis.
Configure the ptime.
Navigate to: http://<phoneIPAddress>/servlet?
p=account-codec&q=load&acc=
0
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Details of the Configuration Parameter:
Parameters
Permitted
Values account.X.codec.Y.enable
(X ranges from 1 to 16. Y ranges from 1 to 11)
0 or 1
Description:
Enables or disables the specified codec for account X.
0-Disabled
1-Enabled
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Default:
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;
Web User Interface:
Account->Codec
Phone User Interface:
None account.X.codec.Y.payload_type
(X ranges from 1 to 16. Y ranges from 1 to 11)
Refer to the following content
Description:
Configures the codec for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Default
Refer to the following content
Refer to the following content
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Configuring Audio Features
Parameters
Permitted
Values
Default
Permitted Values:
PCMU, PCMA, G729, G722, G723_53, G723_63, G726-16, G726-24, G726-32,
G726-40, iLBC
Default:
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;
Example: account.1.codec.1.payload_type = PCMU
Web User Interface:
Account->Codec
Phone User Interface:
None account.X.codec.Y.priority
(X ranges from 1 to 16. Y ranges from 1 to 11)
Integer from 0 to 11
Refer to the following content
Description:
Configures the priority of the enabled codec for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Default:
When Y=1, the default value is 2;
When Y=2, the default value is 3;
When Y=3, the default value is 0;
When Y=4, the default value is 0;
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Parameters
Permitted
Values
Default
When Y=5, the default value is 4;
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;
Web User Interface:
Account->Codec
Phone User Interface:
None account.X.codec.Y.rtpmap
(X ranges from 1 to 16. Y ranges from 1 to 11)
Integer from 0 to 127
Refer to the following content
Description:
Configures the rtpmap of the audio codec for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Default:
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;
Web User Interface:
None
Phone User Interface:
None
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Configuring Audio Features account.X.ptime
Parameters
Permitted
Values
0 (Disabled),
10, 20, 30, 40,
50 or 60
Description:
Configures the ptime (in milliseconds) for the codec for account X.
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Example: account.1.ptime = 20
Web User Interface:
Account->Advanced->PTime(ms)
Phone User Interface:
None
Default
20
To configure the codecs 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 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. Click to remove the codec from the Enable Codecs column.
7. Click or to adjust the priority of the enabled codecs.
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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.
Acoustic Echo Cancellation (AEC) is used to reduce acoustic echo from a voice call to provide natural full-duplex communication patterns. 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. AEC is not normally required for calls via the handset. In certain situation, where echo is experienced by the remote party, AEC may be used to reduce/avoid echo when the user uses the handset.
Configuring Audio Features
Procedure
AEC can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure AEC.
Parameter: voice.echo_cancellation
Configure AEC.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default voice.echo_cancellation 0 or 1 1
Description:
Enables or disables AEC (Acoustic Echo Canceller) feature on the IP phone.
0-Disabled
1-Enabled
Web User Interface:
Settings->Voice->Echo Cancellation->ECHO
Phone User Interface:
None
To configure AEC via web user interface:
1. Click on Settings->Voice.
2. Select the desired value from the pull-down list of ECHO.
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3. Click Confirm to accept the change.
Background noise suppression (BNS) is designed primarily for hands-free operation and reduces background noise to enhance communication in noisy environments.
Automatic Gain Control (AGC) is applicable to hands-free operation and is used to keep audio output at nearly a constant level by adjusting the gain of signals in certain circumstances. This increases the effective user-phone radius and helps with the intelligibility of talkers.
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.
Parameter: voice.vad
Configure VAD.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load
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Configuring Audio Features
Details of the Configuration Parameter:
Parameter Permitted Values Default voice.vad 0 or 1 0
Description:
Enables or disables VAD (Voice Activity Detection) feature on the IP phone.
0-Disabled
1-Enabled
Web User Interface:
Settings->Voice->Echo Cancellation ->VAD
Phone User Interface:
None
To configure VAD via web user interface:
1. Click on Settings->Voice.
2. Select the desired value from the pull-down list of VAD.
3. Click Confirm to accept the change.
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
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Procedure
CNG can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure CNG.
Parameter: voice.cng
Configure CNG.
Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load
Details of the Configuration Parameter:
Parameter Permitted Values Default voice.cng 0 or 1 1
Description:
Enables or disables CNG (Comfortable Noise Generator) feature on the IP phone.
0-Disabled
1-Enabled
Web User Interface:
Settings->Voice->Echo Cancellation ->CNG
Phone User Interface:
None
To configure CNG via web user interface:
1. Click on Settings->Voice.
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2. Select the desired value from the pull-down list of CNG.
Configuring Audio Features
3. Click Confirm to accept the change.
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: fixed and adaptive. A fixed jitter buffer adds the fixed delay to voice packets. You can configure the delay time for the static jitter buffer on IP phones. An adaptive 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.
Parameters: voice.jib.adaptive voice.jib.min voice.jib.max voice.jib.normal
Configure the mode of jitter buffer and the delay time for jitter buffer.
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Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default voice.jib.adaptive 0 or 1 1
Description:
Configures the type of jitter buffer.
0-Fixed
1-Adaptive
Web User Interface:
Settings->Voice->JITTER BUFFER->Type
Phone User Interface:
None voice.jib.min Integer from 0 to 400 60
Description:
Configures the minimum delay time (in milliseconds) of jitter buffer.
Note: It works only if the value of the parameter “voice.jib.adaptive” is set to 1
(Adaptive).
Web User Interface:
Settings->Voice->JITTER BUFFER->Min Delay
Phone User Interface:
None voice.jib.max Integer from 0 to 400 240
Description:
Configures the maximum delay time (in milliseconds) of jitter buffer.
Note: It works only if the value of the parameter “voice.jib.adaptive” is set to 1
(Adaptive).
Web User Interface:
Settings->Voice->JITTER BUFFER->Max Delay
Phone User Interface:
None
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Configuring Audio Features
Parameters Permitted Values Default voice.jib.normal Integer from 0 to 400 120
Description:
Configures the normal delay time (in milliseconds) of jitter buffer.
Note: It works only if the value of the parameter “voice.jib.adaptive” is set to 0
(Fixed).
Web User Interface:
Settings->Voice->JITTER BUFFER->Normal
Phone User Interface:
None
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.
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.
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SIP-T4X IP phones support TLS 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. SIP-T4X IP phones support the following cipher suites:
DHE-RSA-AES256-SHA
DHE-DSS-AES256-SHA
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
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Configuring Security Features
The following figure illustrates the TLS messages exchanged between the IP phone and
TLS server to establish an encrypted communication channel:
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 negotiation with “Server Hello Done” message.
Step3: The 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 will appear 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 up to 10 custom certificates to the IP phone. The format of the certificates 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 installed by default and is unique to an IP phone (based on the MAC address) and issued by the Yealink Certificate
Authority (CA).
- A generic server certificate: It is installed by default and is issued by the
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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.
Common Name Validation feature enables the IP phone to mandatorily validate the common name of the certificate sent by the connecting server.
And Security verification rules are compliant with RFC 2818.
Note
For TLS feature, we use the terms trusted and server certificates. 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.
Procedure
Configuration changes can be performed using the configuration files or locally.
Configuration File
<MAC>.cfg
<y0000000000xx>.cfg
Configure TLS on a per-line basis.
Parameter: account.X.transport
Configure the trusted certificates feature.
Parameters: security.trust_certificates security.ca_cert security.cn_validation
Configure the server certificates feature.
Parameters: security.dev_cert
Upload the trusted certificates.
Parameter: trusted_certificates.url
Upload the server certificates.
Parameter: server_certificates.url
Configure the custom
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Local Web User Interface
Details of Configuration Parameters:
Parameters Permitted Values account.X.transport 0, 1, 2 or 3
Description:
Configures the type of transport protocol for account X.
0-UDP
1-TCP
2-TLS
3-DNS-NAPTR
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Configuring Security Features certificates.
Parameter: phone_setting.reserve_certs_en
able
Configure TLS on a per-line basis.
Navigate to: http://<phoneIPAddress>/servl et?p=account-register&q=load
&acc=0
Configure the trusted certificates feature.
Upload the trusted certificates.
Navigate to: http://<phoneIPAddress>/servl et?p=trusted-cert&q=load
Configure the server certificates feature.
Upload the server certificates.
Navigate to: http://<phoneIPAddress>/servl et?p=server-cert&q=load
Default
0
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Parameters
Account->Register ->Transport
Phone User Interface:
None security.trust_certificates
Permitted Values Default
0 or 1 1
Description:
Enables or disables the IP phone to only trust the server certificates in the Trusted
Certificates list.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the IP phone will authenticate the server certificate based on the trusted certificates list. Only when the authentication succeeds, the IP phone will trust the server.
If it is set to 0 (Disabled),
the IP phone will trust the server no matter whether the certificate sent by the server is valid or not.
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Security->Trusted Certificates->Only Accept Trusted Certificates security.ca_cert 0, 1 or 2 2
Description:
Configures the type of certificates in the Trusted Certificates list for the IP phone to authenticate for TLS connection.
0-Default certificates
1-Custom certificates
2-All certificates
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Security->Trusted Certificates->CA Certificates
Phone User Interface:
None security.cn_validation 0 or 1 0
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Configuring Security Features
Parameters Permitted Values Default
Description:
Enables or disables the IP phone to mandatorily validate the CommonName or
SubjectAltName of the certificate sent by the server.
0-Disabled
1-Enabled
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Security->Trusted Certificates->Common Name Validation
Phone User Interface:
None security.dev_cert 0 or 1 0
Description:
Configures the type of the device certificates for the IP phone to send for TLS authentication
.
0-Default certificates
1-Custom certificates
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Security->Server Certificates->Device Certificates
Phone User Interface:
None trusted_certificates.url
URL within 511 characters
Blank
Description:
Configures the access URL of the custom trusted certificate used to authenticate the connecting server.
Example: trusted_certificates.url = http://192.168.1.20/tc.crt
Note: The certificate you want to upload must be in *.pem, *.crt, *.cer or *.der format.
Web User Interface:
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Parameters Permitted Values
Security->Trusted Certificates->Load trusted certificates file
Phone User Interface:
None server_certificates.url
URL within 511 characters
Default
Blank
Description:
Configures the access URL of the certificate the IP phone sends for authentication.
Example: server_certificates.url = http://192.168.1.20/ca.pem
Note: The certificate you want to upload must be in *.pem or *.cer format.
Web User Interface:
Security->Server Certificates->Load server cer file
Phone User Interface:
None
To configure the trusted certificates feature via web user interface:
1. Click on Security->Trusted Certificates.
2. Select the desired value from the pull-down list of Only Accept Trusted Certificates.
3. Select the desired value from the pull-down list of Common Name Validation.
4. Select the desired value from the pull-down list of CA Certificates.
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5. Click Confirm to accept the change.
A dialog box pops up to prompt that the settings will take effect after reboot.
Configuring Security Features
6. Click OK to reboot the phone.
To configure TLS 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. Select TLS from the pull-down list of the Transport.
4. Click Confirm to accept the change.
To upload a trusted certificate via web user interface:
1. Click on Security->Trusted Certificates.
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2. Click Browse to locate the certificate (*.pem,*.crt, *.cer or *.der) from your local system.
3. Click Upload to upload the certificate.
To configure the server certificates feature via web user interface:
1. Click on Security->Server Certificates.
2. Select the desired value from the pull-down list of Device Certificates.
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3. Click Confirm to accept the change.
A dialog box pops up to prompt that the settings will take effect after reboot.
4. Click OK to reboot the phone.
To upload a server certificate via web user interface:
1. Click on Security->Server Certificates.
Configuring Security Features
2. Click Browse to locate the certificate (*.pem or *.cer) from your local system.
3. Click Upload to upload the certificate.
The dialog box pops up to prompt “Success: The Server Certificate has been loaded!
Rebooting, please wait…”.
Secure Real-Time Transport Protocol (SRTP) encrypts RTP streams during VoIP phone calls to avoid interception and eavesdropping. The parties participating in the call must enable SRTP simultaneously. When this feature is enabled on both phones, the encryption algorithm utilized for the session is negotiated between 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 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NzkyM2FjNzQ2ZDgxYjg0MzQwMGVmMGUxMzdmNWFm a=crypto:3 F8_128_HMAC_SHA1_80 inline:NDliMWIzZGE1ZTAwZjA5ZGFhNjQ5YmEANTMzYzA0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv
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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 feature 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 461 .
Procedure
SRTP can be configured using the configuration files or locally.
Configuration File
Local
<MAC>.cfg
Web User Interface
Configure SRTP feature on a per-line basis.
Parameter: account.X.srtp_encryption account.X.srtp_auth_tag_mode
Configure SRTP feature on a per-line basis.
Navigate to: http://<phoneIPAddress>/servlet
?p=account-adv&q=load&acc=
0
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Details of the Configuration Parameter:
Parameters Permitted Values
Configuring Security Features
Default account.X.srtp_encryption 0, 1 or 2 0
Description:
Configures whether to use voice encryption service for account X.
0-Disabled
1-Optional
2-Compulsory
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 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->RTP Encryption (SRTP)
Phone User Interface:
None account.X.srtp_auth_tag_mode 0, 1 or 2 0
Description:
Configures the encryption algorithm carried in the SIP message when using voice encryption service for account X.
0-AES-80&&AES-32
1-AES-80
2-AES-32
X ranges from 1 to 16 (for SIP-T48G/T46G).
X ranges from 1 to 12 (for SIP-T42G).
X ranges from 1 to 6 (for SIP-T41P).
Web User Interface:
Account->Advanced->SRTP Auth-tag
Phone User Interface:
None
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To configure SRTP 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. Select the desired key type from the pull-down list of SRTP Auth-tag.
6. Click Confirm to accept the change.
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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 supplies 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 symmetric keys (the same or different keys for configuration files) and generates encrypted configuration files with the same file name as before. This tool also encrypts the plaintext 16-character symmetric keys using a fixed key, which is the same as the one built in the IP phone, and generates new files named as <xx_Security>.enc (xx indicates the name of the configuration file, for example, y000000000028_Security.enc for y000000000028.cfg file). This tool generates another new file named as Aeskey.txt to store the plaintext 16-character symmetric keys for each configuration file.
Configuring Security Features
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 supplies a configuration encryption tool (yealinkencrypt) for Linux platform if required. For more information, refer to
Yealink Configuration Encryption Tool User Guide
.
For the 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 IP 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:
When you start the application tool, a file folder named “Encrypted” is created automatically in the directory where the application tool is located.
2. Click Browse to locate configuration file(s) (e.g., y000000000028.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
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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 and the following special characters are also supported: # $ % * + , - . : = ? @ [ ] ^ _ { }
~.
5. Click Encrypt to encrypt the configuration file(s).
6. Click OK.
The target directory will be automatically opened. You can find the encrypted configuration file(s), encrypted key file(s) and an Aeskey.txt file storing plaintext AES key(s).
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Configuring Security Features
Procedure
Encryption method and AES keys can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure the decryption method and AES keys.
Parameters： auto_provision.aes_key_in_file auto_provision.aes_key_16.com auto_provision.aes_key_16.mac auto_provision.update_file_mode
Configure the AES keys.
Navigate to: http://<phoneIPAddress>/servlet?
p=settings-autop&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default auto_provision.aes_key_in_file 0 or 1 0
Description:
Enables or disables the IP phone to decrypt configuration files using the encrypted
AES keys.
0-Disabled
1-Enabled
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
(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.
Web User Interface:
None
Phone User Interface:
None
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Parameters Permitted Values Default auto_provision.aes_key_16.com 16 characters Blank
Description:
Configures the plaintext AES key for decrypting the Common CFG file.
The valid characters contain: 0 ~ 9, A ~ Z, a ~ z and the following special characters are also supported: # $ % * + , - . : = ? @ [ ] ^ _ { } ~.
Example: auto_provision.aes_key_16.com = 0123456789abcdef
Note: It works only if the value of the parameter “auto_provision.aes_key_in_file” is set to 0 (Disabled).
Web User Interface:
Settings->Auto Provision->Common AES Key
Phone User Interface:
None auto_provision.aes_key_16.mac 16 characters Blank
Description:
Configures the plaintext AES key for decrypting the MAC-Oriented CFG file.
The valid characters contain: 0 ~ 9, A ~ Z, a ~ z and the following special characters are also supported: # $ % * + , - . : = ? @ [ ] ^ _ { } ~.
Example: auto_provision.aes_key_16.mac = 0123456789abmins
Note: It works only if the value of the parameter “auto_provision.aes_key_in_file” is set to 0 (Disabled).
Web User Interface:
Settings->Auto Provision->MAC-Oriented AES Key
Phone User Interface:
None auto_provision.update_file_mode 0 or 1 0
Description:
Enables or disables the IP phone to update encrypted configuration settings only during auto provisioning.
0-Disabled
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Configuring Security Features
Parameters
1-Enabled
Web User Interface:
None
Phone User Interface:
None
Permitted Values Default
To configure the AES keys via web user interface:
1. Click on Settings->Auto Provision.
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 and the following special characters are also supported: # $ % * +, - . : = ? @ [ ] ^ _ { } ~.
3. Click Confirm to accept the change.
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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 the IP phone. 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.
The names of the Yealink-supplied template files are (You can rename the filename as required):
Template File
Replace Rule Template
Dial-now Template
Softkey Layout Template dialplan.xml dialnow.xml
File Name
CallFailed.xml
CallIn.xml
Connecting.xml
Dialing.xml (not applicable to
SIP-T48G)
RingBack.xml
Talking.xml
Directory Template
Super Search Template favorite_setting.xml super_search.xml
Local Contact File
Remote XML Phone Book contact.xml
Department.xml
Menu.xml
This chapter provides the detailed information on how to customize the following resource files:
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482
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 file, learn the following:
<DialRule> indicates the start of the template file and </DialRule> indicates the end of the template file.
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.
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 Dial Plan on page 115 .
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 leaving it blank, this replace rule will apply to all lines.
3. Specify the values within double quotes.
4. Place this file to the provisioning server.
Resource Files
The following shows an example of a replace rule file:
<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>
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.
At most 100 dial-now rules can be added to the IP phone.
The expression syntax in the dial-now rule template is the same as that introduced
in the section Dial Plan on page 115 .
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 leaving it blank, this rule will apply to all lines.
3. Specify the values within double quotes.
4. Save the change and place this file to the provisioning server.
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The following shows 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>
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The softkey layout template allows assigning different soft key layouts to different call states. The call states include CallFailed, CallIn, Connecting, Dialing (not applicable to
SIP-T48G), 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>.
<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. Save the change and place this file to the provisioning server.
The following shows an example of the CallFailed template file:
<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>
Resource Files
Directory provides easy access to frequently used lists. Users can access lists by pressing the Directory soft key when the IP phone is idle. The lists may contain Local
Directory, History, Remote Phone Book and LDAP. You can add the desired list(s) to
Directory using the supplied directory template (favorite_setting.xml). 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.
The default display names of directory lists are Local Directory, History, Remote
Phone Book and LDAP.
When specifying the display priority of the directory list, the valid values are 1, 2, 3
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and 4. 1 is the highest priority, 4 is the lowest.
When enabling or disabling the desired directory list for Directory, 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, edit the corresponding string in the file. For example, you want to configure the local directory list, edit the following strings:
<item id_name="localdirectory" display_name="Local Directory" priority="1" enable="1" />
Where: id_name="" specifies the existing directory list (“localdirectory” for the local directory list). Do not edit this field. display_name="" specifies the display name of the directory list. We recommend you do not edit this field. priority="" specifies the display priority of the directory list. enable="" enables or disables the directory list for Directory.
3. Edit the values within double quotes.
4. Place this file to the provisioning server.
The following shows 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>
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Search source list in dialing allows the IP phone to search for entries from the desired lists based on the entered string when in the pre-dialing screen, and then the user can select the desired entry to dial out quickly. The lists may contain Local Directory, History,
Resource Files
Remote Phone Book and LDAP. You can configure the search source list in dialing using the supplied super search template (super_search.xml). 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.
The default display names of directory lists are Local Directory, History, Remote
Phone Book and LDAP.
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, edit the corresponding string in the file. For example, you want to configure the local directory list, edit the following strings:
<item id_name="local_directory_search" display_name="Local Directory" priority="1" enable="1" />
Where: id_name="" specifies the directory list (“local_directory_search” for the local directory list). Do not edit this field. display_name="" specifies the display name of the directory list. We recommend you do not edit this field. priority="" specifies the priority of search results. enable="" enables or disables the IP phone to search the directory list.
3. Edit the values within double quotes.
4. Place this file to the provisioning server.
The following shows an example of a super search template:
<root_super_search>
<item id_name="local_directory_search" display_name="Local
Directory" priority="1" enable="1" />
<item id_name="calllog_search" display_name="History" priority="2" enable="1" />
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<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>
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 file, 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: Silent.wav, Resource: Splash.wav or
Resource: RingN.wav (the default system ring tone, integer N ranges from 1 to 8) or
Custom: Name.wav (the custom ring tone).
When specifying a desired line for a contact, valid values are -1~15. Multiple line
IDs are separated by commas.
The following table lists valid values for each phone model.
Phone Model
SIP-T41P
SIP-T42G
SIP-T46G/T48G
Values
-1~5
-1~11
-1~15
Description
-1 stands for Auto (the first registered line)
0~5 stand for line1~line6
-1 stands for Auto (the first registered line)
0~11 stand for line1~line12
-1 stands for Auto (the first registered line)
0~15 stand for line1~line16
When specifying an avatar for a contact, valid values are Default: avatar name
(the built-in avatar) and Config: avatar name (the custom avatar). It is not applicable to SIP-T42G and SIP-T41P IP phones.
Procedure
Use the following procedures to customize a local contact template file.
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Resource Files
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="" default_photo="" />
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. 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. default_photo=”” specifies the avatar for this contact.
4. Specify the values within double quotes.
5. Save the change and place this file to the provisioning server.
The following shows 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" default_photo="Defult:default_contact_image.png"/>
<contact display_name="Alice" office_number="1008" mobile_number="" other_number="" line="2" ring="Resource:Silent.wav" group_id_name="Friend" default_photo="Config:custom.png"/>
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</root_contact>
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IP phones can access 5 remote phone books. You can customize the remote XML phone book for IP phones as required. You can also add multiple remote contacts at a time and/or share remote contacts between IP phones using the supplied template files
(Menu.xml and Department.xml). The Menu.xml file defines departments of a remote phone book. The Department.xml file defines contact lists for a department, which is nested in Menu.xml file. After setup, place the files (Menu.xml and Department.xml) to the provisioning server, and specify the access URL of the file (Menu.xml) in the configuration files.
When creating a Menu.xml file, learn the following:
<YealinkIPPhoneMenu> indicates the start of a remote phone book file and
</YealinkIPPhoneMenu> indicates the end of a remote phone book file.
Create the title of a remote phone book between <Title> and </Title>.
<MenuItem>indicates the start of specifying a department file and </MenuItem> indicates the end of specifying a department file.
<SoftKeyItem> indicates the start of specifying a XML file and </SoftKeyItem> indicates the end of specifying a XML file.
Procedure
Use the following procedures to customize an XML phone book.
To customize a Menu.xml file:
1. Open the template file using an ASCII editor.
2. For each department that you want to add, add the following strings to the file.
Each starts on a separate line:
<MenuItem>
<Name>
Department1
</Name>
<URL>
http://10.3.6.117:8080/Department1.xml
</URL>
</MenuItem>
Where:
Specify the name of a department between <Name> and </Name>.
Specify the access URL of a department file between </URL> and </URL>.
3. For each XML file that you want to add, add the following strings to the file. Each starts on a separate line:
<SoftKeyItem>
<Name>
#
</Name>
Resource Files
<URL>
http://10.3.6.128:8080/TextMenu.xml
</URL>
</SoftKeyItem>
Where:
Specify the key between <Name> and </Name>.
Specify the access URL of a XML file between </URL> and </URL>.
4. Save the file and place this file to the provisioning server.
The following shows an example of a Menu.xml file:
<YealinkIPPhoneMenu>
<Title>XiaMen Yealink</Title>
<MenuItem>
<Name>Department1</Name>
<URL>http://10.2.9.1:99/Department.xml</URL>
</MenuItem>
<MenuItem>
<Name>Department2</Name>
<URL>http://10.2.9.1:99/Department.xml</URL>
</MenuItem>
<SoftKeyItem>
<Name>#</Name>
<URL>http://10.2.9.1:99/Department.xml</URL>
</SoftKeyItem>
<SoftKeyItem>
<Name>*</Name>
<URL>http://10.2.9.1:99/Department.xml</URL>
</SoftKeyItem>
<SoftKeyItem>
<Name>1</Name>
<URL>http://10.2.9.1:99/Department.xml</URL>
</SoftKeyItem>
</YealinkIPPhoneMenu>
When creating a Department.xml file, learn the following:
<YealinkIPPhoneDirectory> indicates the start of a department file and
</YealinkIPPhoneDirectory> indicates the end of a department file.
Create contact lists for a department between <DirectoryEntry> and
</DirectoryEntry>.
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To customize a Department.xml file:
1. Open the template file using an ASCII editor.
2. For each contact that you want to add, add the following strings to the file. 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. Save the file and place this file to the provisioning server.
The following shows an example of a Department.xml file:
<YealinkIPPhoneDirectory>
<DirectoryEntry>
<Name>Test1</Name>
<Telephone>23000</Telephone>
</DirectoryEntry>
<DirectoryEntry>
<Name>Test2</Name>
<Telephone>303</Telephone>
<Telephone>915980830849</Telephone>
</DirectoryEntry>
<DirectoryEntry>
<Name>Test3</Name>
<Telephone>6650</Telephone>
<Telephone>915980830849</Telephone>
</DirectoryEntry>
</YealinkIPPhoneDirectory>
Note
Yealink supplies a phone book generation tool to quickly generate a remote XML phone book. For more information, refer to
Yealink Phonebook Generation Tool User Guide
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
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Troubleshooting
This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using SIP-T4X 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.
Enabling the Watch Dog Feature
Getting Information from Status Indicators
If your IP phone encounters some problems, commonly the log files are needed. 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.
In the configuration files, you can use the following parameters to configure system log settings:
syslog.mode – Specify the system log to be exported to a server or local system. syslog.server -- Specify the IP address or domain name of the syslog server to which the log will be exported. syslog.log_level -- Specify the system log level. The following lists the log level of events you can log:
0: system is unusable
1: action must be taken immediately
2: critical condition
3: error conditions
4: warning conditions
5: normal but significant condition
6: informational
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Procedure
Log setting can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configures the syslog mode.
Parameters: syslog.mode
Configures the IP address or domain name of the syslog server where to export the log files.
Parameters: syslog.server
Configures the severity level of the logs to be reported to a log file.
Parameters: syslog.log_level
Configures the syslog mode.
Configures the IP address or domain name of the syslog server where to export the log files.
Configures the severity level of the logs to be reported to a log file.
Navigate to: http://<phoneIPAddress>/ servlet?p=settings-config
&q=load
Details of Configuration Parameters:
Parameters Permitted Values Default syslog.mode 0 or 1 0
Description:
Configures the IP phone to export log files to a syslog server or the local system.
0-Local
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Parameters Permitted Values Default
1-Server
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
Settings->Configuration->Export System Log
Phone User Interface:
None syslog.server IP address or domain name Blank
Description:
Configures the IP address or domain name of the syslog server when exporting log to the syslog server.
Example: syslog.server = 192.168.1.30
Note: It works only if the value of 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.
Web User Interface:
Settings->Configuration->Server Name
Phone User Interface:
None syslog.log_level Integer from 0 to 6 3
Description:
Configures the detail level of syslog information to be exported.
0: system is unusable
1: action must be taken immediately
2: critical condition
3: error conditions
4: warning conditions
5: normal but significant condition
6: informational
Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Web User Interface:
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Parameters Permitted Values
Settings->Configuration->System Log Level
Phone User Interface:
None
To configure the system log level via web user interface:
1. Click on Settings->Configuration.
2. Select the desired level from the pull-down list of System Log Level.
Default
3. Click Confirm to accept the change.
The system log level is set as 6, the informational level.
Note
Informational level may make some sensitive information accessible (e.g., password-dial number), we recommend that you reset the system log level to 3 after providing the syslog file.
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.
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Troubleshooting
3. Enter the IP address or domain name of the syslog server in the Server Name field.
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 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. Click Export to open file download window, and then save the file to your local system.
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The following figure shows a portion of a log file:
498
You can capture packets in two ways: capturing the packets via web user interface or using the Ethernet software. You can analyze the packets 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.
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 in the configuration files.
Procedure
Watch Dog can be configured using the configuration files or locally.
Configuration File
Local
<y0000000000xx>.cfg
Web User Interface
Configure Watch Dog feature.
Parameter: watch_dog.enable
Configure Watch Dog feature.
Navigate to: http://<phoneIPAddress>
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Details of the Configuration Parameter:
Parameter Permitted Values
/servlet?p=settings-prefer ence&q=load
Default watch_dog.enable 0 or 1 1
Description :
Enables or disables Watch Dog feature.
0-Disabled
1-Enabled
If it is set to 1 (Enabled), the IP phone will reboot automatically when the system is broken down.
Web User Interface:
Settings->Preference->Watch Dog
Phone User Interface:
None
To configure watch dog via web user interface:
1. Click on Settings->Preference.
2. Select the desired value from the pull-down list of Watch Dog.
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3. Click Confirm to accept the change.
Troubleshooting
Status indicators may consist of the power LED, line key indicator, headset key indicator, mute key indicator and the on-screen icon.
The following shows two examples of obtaining the IP phone information from status indicators on SIP-T46G IP phones:
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 an active call on the IP phone is on mute, the Mute key LED illuminates.
For more information on the icons, refer to Reading Icons on page 14 .
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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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 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 and 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.
If you have poor sound quality/acoustics like intermittent voice, low volume, echo or other noise, 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 jitter, due to message recombination of transmission or receiving equipment (e.g., timeout handling, retransmission mechanism or 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.
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Troubleshooting
A remote phone book is placed on a server, while a local phone book is placed on the 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 message is formed as below:
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
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
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Yealink_SIP-T2_Series_T4_Series_IP_Phones_Auto_Provisioning_Guide
, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142 .
You can press the volume key to increase or decrease the ringer volume and receiver volume. Press the volume key to adjust the ringer volume when the IP phone is idle, or to adjust the volume of the engaged audio device (handset, speakerphone or headset) when there is an active call in progress.
504
IP phones use the PoE preferentially.
Auto provisioning refers to the update of IP phones, including update on configuration parameters, local phonebook, 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 SUBSCRIBE 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 configuration files from the provisioning server. PnP depends on support from a SIP server.
’
Do one of the following:
Ensure that the configuration is set correctly.
Reboot the 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.
Troubleshooting
“ ” “ ”
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.
For anonymous call/anonymous call rejection feature, the phone will send either the on code or off code to the server according to the value of Send Anonymous Code/Send
Rejection Code. For more information, refer to Anonymous Call on page 154 and
Anonymous Call Rejection on page 158 .
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->Advanced (default password: admin) ->Network->WAN Port->IPv4 (or
IPv6). If the Static IP is selected, select DHCP instead.
Reset your phone to factory configurations after you have tried all troubleshooting suggestions but do not resolve the problem. Note that all custom settings will be overwritten after resetting.
To reset your phone via web user interface:
1. Click on Settings->Upgrade.
2. Click Reset to Factory in the Reset to Factory Setting field.
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Administrator’s Guide for SIP-T4X IP Phones
The web user interface prompts the message “Do you want to reset to factory?”.
3. Click OK to confirm the resetting.
The IP phone will be reset to factory sucessfully after startup.
Note
Reset of the phone may take a few minutes. Do not power off until the IP phone starts up successfully.
Factory reset can restore the original password. All custom settings will be overwritten after reset.
Wrong configurations may have an impact on your phone use. You can export configuration files to check the current configuration of the IP phone and troubleshoot if necessary. You can also import configuration files for a quick and easy configuration.
Three types of configuration files can be exported to your local system: config.bin,
<mac>-all.cfg and <mac>-local.cfg. The <mac>-all.cfg configuration file contains all changes made via phone user interface, web user interface and using configuration files. The <mac>-local.cfg configuration file contains changes made via phone user interface and web user interface. The config.bin file is an encrypte file. For more information on config.bin file, contact your Yealink reseller.
To export a BIN configuration file via web user interface:
1. Click on Settings->Configuration.
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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.
To export CFG configuration files via web user interface:
1. Click on Settings->Configuration.
2. Select Local Configuration or All Configuration from the pull-down list of Export CFG
Configuration File and then click Export to save the file to your local system.
To import a BIN configuration file via web user interface:
1. Click on Settings->Configuration.
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Administrator’s Guide for SIP-T4X IP Phones
2. In the Export or Import Configuration block, click Browse to locate a BIN configuration file from your local system.
3. Click Import to import the configuration file.
To import CFG configuration files via web user interface:
1. Click on Settings->Configuration.
2. In the Import CFG Configuration File block, click Browse to locate a CFG configuration file from your local system.
508
3. Click Import to import the configuration file.
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 offers an authentication mechanism for devices to connect 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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Administrator’s Guide for SIP-T4X IP Phones
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.
510
Appendix
−04:00
−04:00
−04:00
−04:00
−04:00
−04:00
−03:30
−03:00
−03:00
−03:00
−03:00
−02:30
−02:00
−01:00
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
−09:30
−09:00
−08:00
−08:00
−08:00
−07:00
Time Zone Name
Samoa
United States-Hawaii-Aleutian
French Polynesia
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)
Newfoundland and Labrador
Brazil(no DST)
Portugal(Azores)
GMT
Greenland
Denmark-Faroe Islands(Torshavn)
Ireland(Dublin)
Portugal(Lisboa, Porto, Funchal)
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+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
Time Zone
0
0
0
+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
+02:00
+03:00
+03:00
+03:00
+03:30
+04:00
+04:00
+04:00
+04:00
Time Zone Name
Spain-Canary Islands(Las Palmas)
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)
512
+07:00
+07:00
+08:00
+08:00
+08:00
+08:00
+08:45
+09:00
+09:00
+09:00
+09:30
+09:30
+10:00
Time Zone
+04:00
+04:30
+05:00
+05:00
+05:00
+05:00
+05:30
+05:45
+06:00
+06:00
+06:30
+10:00
+10:00
+10:00
+10:30
+11:00
+11:00
+11:30
+12:00
+12:00
+12:45
+13:00
+13:30
+14:00
Time Zone Name
Russia(Samara)
Afghanistan(Kabul)
Kazakhstan(Aqtobe)
Kyrgyzstan(Bishkek)
Pakistan(Islamabad)
Russia(Chelyabinsk)
India(Calcutta)
Nepal(Katmandu)
Kazakhstan(Astana, Almaty)
Russia(Novosibirsk, Omsk)
Myanmar(Naypyitaw)
Russia(Krasnoyarsk)
Thailand(Bangkok)
China(Beijing)
Singapore(Singapore)
Australia(Perth)
Russian(Irkutsk, Ulan-Ude)
Eucla
Korea(Seoul)
Japan(Tokyo)
Russian(Yakutsk, Chita)
Australia(Adelaide)
Australia(Darwin)
Australia(Sydney, Melbourne, Canberra)
Australia(Brisbane)
Australia(Hobart)
Russia(Vladivostok)
Australia(Lord Howe Islands)
New Caledonia(Noumea)
Russia(Srednekolymsk Time)
Norfolk Island
New Zealand(Wellington, Auckland)
Russian(Kamchatka Time)
New Zealand(Chatham Islands)
Tonga(Nukualofa)
Chatham Islands
Kiribati
Appendix
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Administrator’s Guide for SIP-T4X IP Phones
514
This appendix describes the DSS key parameters you can configure on IP phones. DSS keys can be assigned with various key features. The DSS key consists of line key and programable key. The following table lists the number of DSS keys you can configure for each phone model:
Parameter- linekey.X.type
Parameter- programablekey.X.type
Configuration File
<y0000000000xx>.cfg
Description
Configures the key feature for the DSS key.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
For line keys:
Valid types are:
N/A
Conference
Forward
Transfer
Hold
DND
ReCall
SMS (not applicable to
SIP-T48G/T42G/T41P)
Directed Pickup
Call Park
DTMF
Voice Mail
Speed Dial
Intercom
Appendix
Line
BLF
URL
Group Listening
Hot Desking
XML Group
Group Pickup
Multicast Paging
Record
XML Browser
URL Record
LDAP
Prefix
Zero Touch
ACD
Local Group
Phone Lock(not applicable to SIP-T48G)
Directory
For programable keys:
Valid types are:
N/A
Forward
DND
ReCall
SMS (not applicable to
SIP-T48G/T42G/T41P)
Directed Pickup
Spead Dial
XML Group
Group Pickup
XML Browser
History
Menu
New SMS (not applicable to
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Administrator’s Guide for SIP-T4X IP Phones
Format
Default Value
SIP-T48G/T42G/T41P)
Status
Hot Desking (only applicable to
SIP-T48G/T46G)
LDAP
Prefix
Local Directory
Local Group
XML Directory
Phone Lock (not applicable to
SIP-T48G)
Switch Account Up
Switch Account Down
Directory
Integer
For line keys:
For SIP-T46G IP phones:
The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is
0.
For SIP-T42G IP phones:
The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is
0.
For SIP-T41P IP phones:
The default value of the line key 1-6 is 15, and the default value of the line key 7-15 is
0.
For programable keys:
For
SIP-T46G
IP phones:
When X=1, the default value is 28 (
History
).
When X=2, the default value is 61
(
Directory
).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 ( History ).
When X=6, the default value is 61
516
Range
Appendix
( Directory ).
When X=7, the default value is 51 (
Switch
Account Up
).
When X=8, the default value is 52 (
Switch
Account Down ).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 ( NA ).
When X=12, the default value is 0 (
NA
).
When X=13, the default value is 0 (
NA
).
When X=14, the default value is 2
(
Forward
).
For
SIP-T42G/T41P
IP phones:
When X=1, the default value is 28 ( History ).
When X=2, the default value is 61
(
Directory
).
When X=3, the default value is 5 (
DND
).
When X=4, the default value is 30 (
Menu
).
When X=5, the default value is 28 (
History
).
When X=6, the default value is 61
( Directory ).
When X=7, the default value is 51 ( Switch
Account Up
).
When X=8, the default value is 52 (
Switch
Account Down
).
When X=9, the default value is 33 (Status).
When X=10, the default value is 0 ( NA ).
When X=13, the default value is 0 ( NA ).
Valid values are:
0-N/A
1-Conference
2-Forward
3-Transfer
4-Hold
5-DND
7-ReCall
8-SMS (not applicable to
SIP-T48G/T42G/T41P)
9-Directed Pickup
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Administrator’s Guide for SIP-T4X IP Phones
Example
10-Call Park
11-DTMF
12-Voice Mail
13-SpeedDial
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
32-New SMS (not applicable to
SIP-T48G/T42G/T41P)
33-Status
34-Hot Desking
35-URL Record
38-LDAP
40-Prefix
41-Zero Touch
42-ACD
43-Local Directory
45-Local Group
47-XML Directory
50-Phone Lock (not applicable to SIP-T48G)
51-Switch Account Up
52-Switch Account Down
61-Directory linekey.1.type = 15
518
Parameter- linekey.X.line
Parameter- programablekey.X.line
Description
Appendix
Configuration File
<y0000000000xx>.cfg
Configures the desired line to apply the key feature.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
When assigning the following features, you don’t need to configure this parameter:
DTMF
Prefix
XML Browser
LDAP
Conference
Forward
Hold
DND
ReCall
SMS (not applicable to
SIP-T48G/T42G/T41P)
Record
URL Record
Multicast Paging
Group Listening
Local Group
XML Group
Zero Touch
URL
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Administrator’s Guide for SIP-T4X IP Phones
Format
Default Value
Range
Example
Parameter- linekey.X.value
Parameter- programablekey.X.value
ACD
Hot Desking
Phone Lock
Directory
Integer
For programable keys, the default value is not applicable.
For line keys, 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 16 (for SIP-T48G/T46G)
1 to 12 (for SIP-T42G)
1 to 6 (for SIP-T41P)
1-Line 1
2-Line 2
…
16-Line 16 linekey.1.line = 2
Configuration File
<y0000000000xx>.cfg
Description
Configures the value for some key features.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
String
Blank
520
Format
Default Value
Range
Example
Parameter- linekey.X.label
Parameter- programablekey.X.label
Description
Format
Default Value
Range
Example
Parameter- linekey.X.pickup_value
Description
Format
Default Value
Range
Appendix
String within 99 characters
When you assign the Speed Dial to the line key, this parameter is used to specify the number you want to dial out. linekey.1.value = 1001
Configuration File
<y0000000000xx>.cfg
Configures the label displaying on the LCD screen for each line key and each soft key.
This is an optional configuration.
For line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys, x ranges from 1 to 4.
String
Blank
String within 99 characters linekey.1.label = Dir
Configuration File
<y0000000000xx>.cfg
Configures the pickup code for BLF feature.
This parameter is only applicable to BLF feature.
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
String
Blank
String within 256 characters
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Administrator’s Guide for SIP-T4X IP Phones
Example linekey.1.pickup_value = *88
Parameter- linekey.X.xml_phonebook
Parameter- programablekey.X.xml_phonebook
Configuration File
<y0000000000xx>.cfg
Description
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 line keys:
X ranges from 1 to 29 (for SIP-T48G).
X ranges from 1 to 27 (for SIP-T46G).
X ranges from 1 to 15 (for SIP-T42G/T41P).
For programable keys:
X=1-10, 12-14 (for SIP-T48G/T46G)
X=1-10, 13 (for SIP-T42G/T41P)
When the key feature is configured as
Local Group, valid values are:
0-All contacts
1-First local group
…
48-
Forty-eighth local group
Format
Default Value
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
Range
Example
0 to 48
Configures the second remote phone
522
book. linekey.1.xml_phonebook = 1
Appendix
This section describes how Yealink SIP-T4X 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
The following RFC’s and Internet drafts are supported:
RFC 1321—The MD5 Message-Digest Algorithm
RFC 1889—RTP Media control
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 2976—The SIP INFO Method
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)
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Administrator’s Guide for SIP-T4X IP Phones
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
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)
524
Appendix
RFC 3863—Presence Information Data Format
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 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.
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
RFC 5763—Framework for Establishing a Secure Real-time Transport Protocol (SRTP)
RFC 5806—Diversion Indication in SIP 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-00.txt—SIP Extensions for Caller Identity and Privacy,
November draft-ietf-sip-privacy-04.txt—SIP Extensions for Network-Asserted Caller Identity
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Administrator’s Guide for SIP-T4X IP Phones
and Privacy within Trusted Networks draft-ietf-sipping-cc-conferencing-03.txt—SIP Call Control - Conferencing for User
Agents draft-ietf-sip-connect-reuse-06.txt—Connection Reuse in the Session Initiation
Protocol (SIP) draft-ietf-bliss-shared-appearances-15.txt—Shared Appearances of a Session
Initiation Protocol (SIP) Address of Record (AOR) draft-anil-sipping-bla-04.txt—Implementing Multiple Line Appearances using the
Session Initiation Protocol (SIP)
To find the applicable Request for Comments (RFC) document, go to http://www.ietf.org/rfc.html
and enter the RFC number.
526
The following SIP request messages are supported:
Method
REGISTER
Supported
Yes
INVITE
ACK
CANCEL
BYE
OPTIONS
SUBSCRIBE
NOTIFY
REFER
PRACK
INFO
MESSAGE
UPDATE
PUBLISH
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Notes
Yealink SIP-T4X IP phones support mid-call changes such as putting a call on hold as signaled by a new
INVITE that contains an existing Call-ID.
Appendix
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
P-Preferred-Identity
Proxy-Authenticate
Proxy-Authorization
RAck
Record-Route
Refer-To
Referred-By
Remote-Party-ID
Replaces
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Supported
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Notes
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Require
Route
Method
RSeq
Session-Expires
Subscription-State
Supported
To
User-Agent
Via
Supported
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Notes
528
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
2xx Response
200 OK
202 Accepted
Supported
Yes
Yes
3xx Response—Redirection Responses
3xx Response
300 Multiple Choices
301 Moved Permanently
Supported
Yes
Yes
Notes
Notes
In REFER transfer.
Notes
3xx Response
302 Moved Temporarily
Supported
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
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
Yes
No
No
No
No
Yes
Yes
No
No
No
Yes
Yes
Yes
Supported
Yes
Yes
Yes
Yes
Yes
Yes
No
Yes
Yes
No
Yes
No
Notes
Notes
Appendix
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4xx Response
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
491 Request Pending
493 Undecipherable
Supported
Yes
Yes
Yes
No
No
5xx Response—Server Failure Responses
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
Notes
Notes
Notes
530
SDP Headers v—Protocol version o—Owner/creator and session identifier a—Media attribute c—Connection information
Supported
Yes
Yes
Yes
Yes
Appendix
SDP Headers m—Media name and transport address s—Session name t—Active time
Supported
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
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.
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3. User B hangs up.
User A Proxy Server
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
F14. 200 OK
User B
Step Action
F1
INVITE—User A to Proxy
Server
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 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.
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Appendix
Step
F2
F3
F4
F5
F6
F7
F8
F9
F10
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
ACK—User A to Proxy Server
ACK—Proxy Server to User B
Description
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 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
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Step
F11
F12
F13
F14
200 OK—User A to Proxy
Server
Action
BYE—User B to Proxy Server
BYE—Proxy Server to User A
200 OK—Proxy Server to User
B
Description server has received the 200 OK response. The call session is now active.
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.
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.
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Appendix
The call cannot be set up successfully.
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
F8. ACK
Step
F1
F2
INVITE—User A to Proxy
Server
Action
INVITE—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. Proxy server forwards the INVITE message to User B.
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Administrator’s Guide for SIP-T4X IP Phones
Step
F3
F4
F5
F6
F7
F8
Action
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
Description
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.
536
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.
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.
Appendix
The call cannot be set up successfully.
User A Proxy Server User B
F1. INVITE B
F2. INVITE B
F3. 180 Ringing
F4. 180 Ringing
F5. CANCEL
F6. CANCEL
F7. 200 OK
F8. 200 OK
Step
F1
F2
INVITE—User A to Proxy
Server
Action
INVITE—Proxy Server to User
B
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
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Administrator’s Guide for SIP-T4X IP Phones
Step
F3
F4
F5
F6
F7
F8
Action Description the INVITE message to User B.
180 Ringing—User B to Proxy
Server
180 Ringing—Proxy Server to
User A
CANCEL—User A to Proxy
Server
CANCEL—Proxy Server 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
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.
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Appendix
3. User A puts User B on hold.
User A Proxy Server User B
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
F9. INVITE B (sendonly)
F10. INVITE B (sendonly)
F11. 200 OK
F12. 200 OK
F13. ACK
No RTP packets being sent
F14. ACK
Step Action
F1
INVITE—User A to Proxy
Server
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.
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Administrator’s Guide for SIP-T4X IP Phones
Step
F2
F3
F4
F5
F6
F7
F8
F9
Action
Description
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
180 Ringing—User B to Proxy
Server
180 Ringing—Proxy Server to
User A
200 OK—User B to Proxy
Server
200 OK—Proxy Server to User
A
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 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 200 OK message to User A. The 200 OK response notifies User A that the connection has been made.
ACK—User A to Proxy Server
ACK—Proxy Server to User B
INVITE—User A to Proxy
Server
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
540
Appendix
Step
F10
F11
F12
F13
F14
Action
INVITE—Proxy Server to User
B
ACK—User A to Proxy Server
ACK—Proxy Server to User B
Description place the call on hold.
The proxy server forwards the mid-call
INVITE message to User B.
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 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 put 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.
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.
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Administrator’s Guide for SIP-T4X IP Phones
4. User B accepts the call from User C.
User A
Proxy Server
F1. INVITE B
F4. 180 Ringing
F6. 200 OK
F2. INVITE B
F3. 180 Ringing
F5. 200 OK
F7. ACK
F8. ACK
2-way RTP channel established
User B
F9. INVITE A
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
F22. ACK
F12. 180 Ringing
F20. 200 OK
F21. ACK
2-way RTP channel established
User C
Step
F1
Action
INVITE—User A to Proxy
Server
Description
User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a
542
Appendix
Step
F2
F3
F4
F5
F6
F7
Action Description 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.
INVITE—Proxy Server to User
B
180 Ringing—User B to Proxy
Server
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.
180 Ringing—Proxy Server to
User A
200 OK—User B to Proxy
Server
200 OK—Proxy Server to User
A
ACK—User A to Proxy Server
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.
The proxy server forwards the 200 OK message 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
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Administrator’s Guide for SIP-T4X IP Phones
Step
F8
F9
F10
F11
F12
F13
INVITE—User C to Proxy
Server
INVITE—Proxy Server to User
A
Action
ACK—Proxy Server to User B
180 Ringing—User A to Proxy
Server
180 Ringing—Proxy Server to
User C
INVITE—User A to Proxy
Description 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.
User A sends a mid-call INVITE request
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Appendix
Step
F14
F15
F16
F17
F18
F19
F20
F21
F22
Server
INVITE—Proxy Server to User
B
200 OK—User B to Proxy
Server
Action
200 OK—Proxy Server to User
A
Description 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.
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 put on hold.
ACK—User A to Proxy Server
ACK—Proxy Server to User B
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 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 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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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.
4. User C answers the call.
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Appendix
Call is established between User A and User C.
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
Step Action
F1
INVITE—User A to Proxy
Server
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
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Administrator’s Guide for SIP-T4X IP Phones
Step
F2
F3
F4
F5
F6
F7
F8
Action
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.
INVITE—Proxy Server to User
B
180 Ringing—User B to Proxy server
180 Ringing—Proxy Server to
User A
200 OK—User B to Proxy
Server
200 OK—Proxy Server to User
A
ACK—User A to Proxy Server
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 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 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.
ACK—Proxy Server to User B The proxy server sends the SIP ACK to
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Appendix
Step
F9
F10
F11
F12
F13
F14
F15
F16
F17
F18
F19
Action Description
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.
202 Accepted—Proxy Server to User B
REFER—Proxy Server to User
A
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.
The proxy server forwards the REFER message to User A.
202 Accepted—User A to
Proxy Server
BYE—User B to Proxy Server
BYE—Proxy Server to User A
200OK—User A to Proxy
Server
200OK—Proxy Server to User
B
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
C
180 Ringing—User C to Proxy
Server
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.
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 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
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Administrator’s Guide for SIP-T4X IP Phones
Step
F20
F21
F22
F23
F24
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
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.
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.
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Appendix
Call is established between User B and User C.
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-T4X IP Phones
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 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
180 Ringing—User B to Proxy
Server
180 Ringing—Proxy Server to
User A
200 OK—User B to Proxy
Server
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 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
552
Appendix
Step
F7
F8
F9
F10
F11
F12
F13
F14
F15
INVITE—User A to Proxy
Server
Action
ACK—User A to Proxy Server
ACK—Proxy Server to User B
Description 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.
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
The proxy server forwards the mid-call
INVITE message 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 was successfully processed.
The proxy server forwards the 200 OK response to User A. The 200 OK response notifies User B is successfully put on hold.
ACK—User A to Proxy Server
ACK—Proxy Server to User B
INVITE—User A to Proxy
Server
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.
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Administrator’s Guide for SIP-T4X IP Phones
Step
F16
F17
F18
F19
F20
F21
F22
F23
F24
F25
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
REFER—User A to Proxy
Server
Action
ACK— User A to Proxy Server
ACK—Proxy Server to User C
Description
The proxy server maps the SIP URI in the
To field to User C. The proxy server 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.
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.
202 Accepted—Proxy Server to User A
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.
554
Appendix
Step
F26
F27
F28
F29
F30
200OK—User B to Proxy
Server
Action
202 Accepted—User B to
Proxy Server
BYE—User A to Proxy Server
BYE—Proxy Server to User B
200OK—Proxy Server to User
A
Description
User B sends a SIP 202 Accept response to the proxy server. The 202 Accepted 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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Administrator’s Guide for SIP-T4X IP Phones
4. User C answers the call.
Call is established between User A and User C.
User A Proxy Server User B
F1. INVITE B
F2. INVITE B
F3. 302 Move Temporarily
F4. ACK
F5. 302 Move Temporarily
F6. ACK
F7. INVITE C
F8. INVITE C
F9. 180 Ringing
F10. 180 Ringing
F11. 200 OK
F12. 200 OK
F13. ACK
F14. ACK
2-way RTP channel established
Step
F1
INVITE—User A to Proxy
Server
Action
User C
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
556
Appendix
Step
F2
F3
F4
F5
F6
F7
F8
F9
F10
Action
Description
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
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
C
180 Ringing—User C to Proxy
Server
180 Ringing—Proxy Server to
User A
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.
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
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Administrator’s Guide for SIP-T4X IP Phones
Step
F11
F12
F13
F14
200OK—User C to Proxy
Server
Action
200OK—Proxy Server to User
A
ACK—User A to Proxy Server
ACK—Proxy Server to User C
Description 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.
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.
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Appendix
User A
Call is established between User A and User C.
Proxy Server User B
F1. INVITE B
F2. INVITE B
F3. 180 Ringing
F4. 180 Ringing
F5. 302 Move Temporarily
F6. ACK
F7. 302 Move Temporarily
F8. ACK
F9. INVITE C
F10. INVITE C
F11. 180 Ringing
F12. 180 Ringing
F13. 200 OK
F14. 200 OK
F15. ACK
F16. ACK
2-way RTP channel established
Step
F1
INVITE—User A to Proxy
Server
Action
User C
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.
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Administrator’s Guide for SIP-T4X IP Phones
Step
F2
F3
F4
F5
F6
F7
F8
F9
F10
Action
Description
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
180 Ringing—User B to Proxy
Server
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.
180 Ringing—Proxy Server to
User A
302 Move Temporarily—User
B to Proxy Server
ACK—Proxy Server to User B
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
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
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.
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Appendix
Step
F11
F12
F13
F14
F15
F16
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
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.
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 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.
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Administrator’s Guide for SIP-T4X IP Phones
Call is established between User A and User C.
User A Proxy Server User B
F1. INVITE B
F2. INVITE B
F3. 180 Ringing
F4. 180 Ringing
F5. 302 Move Temporarily
F6. ACK
F7. 302 Move Temporarily
F8. ACK
F9. INVITE C
F10. INVITE C
F11. 180 Ringing
F12. 180 Ringing
F13. 200 OK
F14. 200 OK
F15. ACK
F16. ACK
2-way RTP channel established
Step
F1
INVITE—User A to Proxy
Server
Action
User C
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.
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Appendix
Step
F2
F3
F4
F5
F6
F7
F8
F9
F10
Action
Description
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
180 Ringing—User B to Proxy
Server
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.
180 Ringing—Proxy Server to
User A
302 Move Temporarily—User
B to Proxy Server
ACK—Proxy Server to User B
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
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
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.
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Step
F11
F12
F13
F14
F15
F16
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
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 sends the ACK message to User C. The ACK confirms that the proxy server has received the
200 OK response.
564
The following figure illustrates successful 3-way calling between Yealink SIP-T4X 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 puts User B on hold.
4. User A calls User C.
5. User C answers the call.
Appendix
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
Step
F1
INVITE—User A to Proxy
Server
Action
User C
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.
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Administrator’s Guide for SIP-T4X IP Phones
Step
F2
F3
F4
F5
F6
F7
F8
Action
Description
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
180 Ringing—User B to Proxy
Server
180 Ringing—Proxy Server to
User A
ACK—User A to Proxy Server
ACK—Proxy Server to User B
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.
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 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 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.
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Appendix
Step
F9
F10
F11
F12
F13
F14
F15
F16
F17
F18
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 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 put 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.
INVITE—User A to Proxy
Server
INVITE—Proxy Server to User
C
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 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
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Administrator’s Guide for SIP-T4X IP Phones
Step
F19
F20
F21
F22
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
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.
568
Numeric
A
Acoustic Echo Cancellation
Analyzing the Configuration Files 501
Appendix C: Configuring DSS Key 514
Appendix E: SIP Call Flows 531
Audio Codecs
Automatic Call Distribution 297
B
Index
C
Call Display
Call Log
Call Recording
Calling Line Identification Presentation 238
Connected Line Identification Presentation 240
Capturing the Current Screen of the Phone 358
Comfort Noise Generation
Configuring Advanced features 259
Connect the Network and Power 11
Configuring Security Features
D
Dual Headset
E
Encrypting Configuration Files 474
569
Administrator’s Guide for SIP-T4X IP Phones
Enabling the Watch Dog Feature
F
Feature Key Synchronization
G
Getting Information from Status Indicators 501
N
Network Address Translation (NAT) 397
Network Conference
H
Headset Prior
Hide Features Access Code
Hot Desking
O
I
Initialization Process Overview 12
Input Method Customization
Intercom
IPv6 Support
M
Message Waiting Indicator
Multicast Paging
P
Physical Features of SIP-T4X IP Phones 4
Q
J
K
Key Features of SIP-T4X IP Phones
R
Reading Icons
RFC and Internet Draft Support
L
Local Contact File
S
Search Source List in Dialing 133
Semi-attended Transfer
570
Server Redundancy
SIP
SIP Components
SIP Header
SIP Request
SIP Responses
SIP Session Description Protocol Usage
Specifying the Language to Use 95
SRTP
Suppressing DTMF Display
W
Wallpaper
Web User Interface
T
Time and Date
Transfer on Conference Hang Up 211
Transport Layer Security (TLS)
Troubleshooting Methods
TR-069 Device Management
U
Use Outbound Proxy in Dialog 177
User Password
V
Voice Activity Detection
VPN
Voice Quality Monitoring
Index
571
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