Yealink SIP-T48G Administrator's Manual
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Yealink SIP-T48G Administrator's Manual

T2 series; t4 series
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Summary of Contents for Yealink SIP-T48G

  • Page 1 啊...
  • Page 2 Copyright © 2015 YEALINK NETWORK TECHNOLOGY Copyright © 2015 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.
  • Page 3 Do not dispose of WEEE as unsorted municipal waste and have to collect such WEEE separately. We are striving to improve our documentation quality and we appreciate your feedback. Email your opinions and comments to DocsFeedback@yealink.com.
  • Page 4 Yealink IP phone firmware contains third-party software under the GNU General Public License (GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms and conditions of the license. The original GPL license, source code of components licensed under GPL and used in Yealink products can be downloaded from Yealink web site: http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293.

  • Page 5: About This Guide

    IP phone’s performance in the network. So an understanding of IP networking and a prior knowledge of IP telephony concepts are necessary. This guide covers SIP-T48G, SIP-T46G, T42G, T41P , T29G, T27P , T23P/G and T21(P) E2 IP phones. The following related documents are available: Quick Start Guides, which describe how to assemble IP phones and configure the ...
  • Page 6 RFC 3261, SIP call flows and the sample configuration files. This section describes the changes to this guide for each release and guide version. This version is updated to incorporate SIP-T48G IP phones. Documentations of the newly released SIP-T27P and SIP-T21(P) E2 IP phones have also been added.
  • Page 7 About This Guide Hide Features Access Code on page  Major updates have occurred to the following sections: DHCP on page  Call Display on page  Input Method Customization on page  BLF List on page  IPv6 Support on page ...
  • Page 8 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones viii...

  • Page 9: Table Of Contents

    Table of Contents About This Guide ..............v Documentations ..........................v In This Guide ............................ v Summary of Changes ........................vi Changes for Release 80, Guide Version 80.20 ..............vi Changes for Release 80, Guide Version 80.6 ................ vi Table of Contents ..............ix Product Overview ..............
  • Page 10 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Configuring Basic Features ............ 57 Power Indicator LED ........................58 Notification Popups ........................62 Contrast ............................65 Wallpaper ............................66 Backlight ............................70 Call Display ............................ 74 Web Server Type..........................77 User Password ..........................80 Administrator Password ........................

  • Page 11: Table Of Contents

    Table of Contents Use Outbound Proxy in Dialog ....................196 SIP Session Timer ......................... 198 Session Timer ..........................200 Call Hold ............................203 Call Forward ..........................208 Call Transfer ..........................227 Network Conference ........................230 Feature Key Synchronization ...................... 232 Transfer on Conference Hang Up ....................
  • Page 12 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones LLDP ............................... 407 CDP ............................... 411 VLAN ............................. 413 VPN ..............................421 Voice Quality Monitoring ......................424 RTCP-XR ..........................424 VQ-RTCPXR ..........................425 Quality of Service ........................442 Network Address Translation ..................... 445 802.1X Authentication ......................... 449 TR-069 Device Management ......................
  • Page 13 Table of Contents Viewing Log Files ........................523 Capturing Packets ........................ 533 Enabling Watch Dog Feature ....................534 Getting Information from Status Indicators ................ 535 Analyzing Configuration File ....................535 Troubleshooting Solutions ......................538 Why is the LCD screen blank? ..................... 538 Why doesn’t the IP phone get an IP address? ..............
  • Page 14 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Successful Call Setup and Call Hold .................. 577 Successful Call Setup and Call Waiting ................580 Call Transfer without Consultation ..................584 Call Transfer with Consultation .................... 589 Always Call Forward ......................594 Busy Call Forward ........................ 597 No Answer Call Forward .....................

  • Page 15: Product Overview

    Product Overview This chapter contains the following information about IP phones: VoIP Principle  SIP Components  SIP IP Phone Models  VoIP VoIP (Voice over Internet Protocol) is a technology using the Internet Protocol instead of traditional Public Switch Telephone Network (PSTN) technology for voice communications.
  • Page 16 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones SIP provides capabilities to: Determine the location of the target endpoint -- SIP supports address resolution,  name mapping, and call redirection. Determine media capabilities of the target endpoint -- Via Session Description  Protocol (SDP), SIP determines the “lowest level”...
  • Page 17 SIP PBXs. SIP-T48G, SIP-T46G, SIP-T42G, SIP-T41P , SIP-T29G, SIP-T27P , SIP-T23P/G and SIP-T21(P) E2 IP phones provide a powerful and flexible IP communication solution for Ethernet TCP/IP networks, delivering excellent voice quality.

  • Page 18: Physical Features Of Ip Phones

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones For a list of key features available on Yealink IP phones running the latest firmware, refer Key Features of IP Phones on page 12. In order to operate as SIP endpoints in your network successfully, IP phones must meet the following requirements: A working IP network is established.
  • Page 19 Product Overview 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 SIP-T46G Physical Features: 4.3”...
  • Page 20 Administrator’s Guide for SIP-T2_Series_T4_Series 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...
  • Page 21 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...
  • Page 22 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones SIP-T29G 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 41 keys including 10 line keys 1*RJ9 (4P4C) handset port 1*RJ9 (4P4C) headset port 2*RJ45 10/100/1000Mbps Ethernet ports...
  • Page 23 Product Overview SIP-T27P Physical Features: 240x120 graphic LCD 6 VoIP accounts, Broadsoft Validated/Asterisk ® Compatible HD Voice: HD Codec, HD Handset, HD Speaker 39 keys including 8 line keys 1*RJ9 (4P4C) handset port 1*RJ9 (4P4C) headset port 2*RJ45 10/100Mbps Ethernet ports 1*RJ12 (6P6C) expansion module port 11*LEDs: 1*power, 8*line, 1*headset, 1*message Power adapter: AC 100~240V input and DC 5V/1.2A output...
  • Page 24 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones SIP-T23P/G Physical Features: 132x64 graphic LCD with 4-level grayscales 3 VoIP accounts, Broadsoft Validated/Asterisk ® Compatible HD Voice: HD Codec, HD Handset, HD Speaker 31 keys including 4 soft keys 1*RJ9 (4P4C) handset port 1*RJ9 (4P4C) headset port 2*RJ45 10/100/1000Mbps Ethernet ports (1000Mbps is only applicable to SIP-T23G IP phones)
  • Page 25 Product Overview SIP-T21(P) E2 Physical Features: 132x64 graphic LCD 2 VoIP accounts 30 keys including 4 soft keys 4 LEDs: 1*power, 2*line, 1*message HD Voice: HD Codec, HD Handset, HD Speaker 1xRJ9 (4P4C) handset port 1xRJ9 (4P4C) headset port 2xRJ45 10/100Mbps Ethernet ports Power adapter: AC 100~240V input and DC 5V/600mA output Power over Ethernet (IEEE 802.3af) (not applicable to SIP-T21 E2 IP phones) Wall Mount...
  • Page 26 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones In addition to physical features introduced above, IP phones also support the following key features when running the latest firmware: Phone Features  Call Options: emergency call, call waiting, call hold, call mute, call forward, call transfer, call pickup, conference.
  • Page 27 Product Overview SRTP (RFC3711) Transport Layer Security (TLS) VLAN (802.1q), QoS Digest authentication using MD5/MD5-sess Secure configuration file via AES encryption Phone lock for personal privacy protection Admin/User configuration mode...
  • Page 28 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones...

  • Page 29: Getting Started

    Configuring Basic Network Parameters  Upgrading Firmware  This section introduces how to install SIP-T48G/T46G/T42G/T41P/T29G/T27P/T23P/T23G/T21(P) E2 IP phones with components in packaging contents. Attach the stand and the optional wall mount bracket Connect the handset and optional headset Connect the network and power Note A headset, wall mount bracket are not included in packaging contents.
  • Page 30 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Attach the stand and the optional wall mount bracket: For SIP-T46G: Desk Mount Method Wall Mount Method (Optional)
  • Page 31 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...
  • Page 32 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones For SIP-T29G/T27P: Desk Mount Method Wall Mount Method (Optional) For SIP-T23P/T23G/T21(P) E2: Desk Mount Method Wall Mount Method (Optional)
  • Page 33 Bluetooth on SIP-T48G/T46G/T29G IP phones, refer to USB Dongle BT40 User Guide The EXT port on SIP-T48G and SIP-T46G IP phones can also be used to connect the expansion module EXP40. The EXT port on SIP-T29G/T27P IP phones can also be used to connect the expansion module EXP39.
  • Page 34 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones For SIP-T23P/T23G/T21(P) E2: Connect the network and power: AC power (Optional)  Power over Ethernet (PoE)  Note PoE is not applicable to the SIP-T21 E2 IP phones. AC Power (Optional) To connect the AC power and network: Connect the DC plug of the power adapter to the DC5V port on the IP phone and connect the other end of the power adapter into an electrical power outlet.
  • Page 35 Connect the Ethernet cable between the Internet port on the IP phone and an available port on the in-line power switch/hub. For SIP-T48G/T46G/T42G/T41P/T29G/T27P/T23P/T23G/T21P E2 IP phones: Note If in-line power switch/hub is provided, you don’t need to connect the phone to the power adapter.
  • Page 36 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Querying the DHCP (Dynamic Host Configuration Protocol) Server The IP phone is capable of querying a DHCP server. DHCP is enabled on the IP phone by default. The following network parameters can be obtained from the DHCP server during initialization: IP Address ...
  • Page 37 Getting Started The message “Welcome Initializing… please wait” appears on the LCD screen when the IP phone starts up. The main LCD screen displays the following: Time and date  Soft key labels  Press the OK key to check the IP phone status, the LCD screen displays the valid IP address, MAC address, firmware version, etc.
  • Page 38 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones T42G/ T23P/T23G/T2 T48G T46G T29G T27P Description T41P 1(P) E2 Auto Answer Do Not Disturb Call Forward Call Hold Call Mute Ringer volume is Phone Lock Multi-lingual lowercase letters input mode Multi-lingual uppercase letters input mode Alphanumeric input mode...

  • Page 39: Phone User Interface

    Getting Started T42G/ T23P/T23G/T2 T48G T46G T29G T27P Description T41P 1(P) E2 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...

  • Page 40: Configuration Files

    Access to specific features is restricted to the administrator. The default password is “admin“(case-sensitive). Not all features are available on phone user interface. 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”...
  • Page 41 This entire process is called auto provisioning. For more information on auto provisioning, refer to Yealink IP Phones Auto Provisioning Guide, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.

  • Page 42: Provisioning Server

    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 IP Phones Auto Provisioning Guide.

  • Page 43: Deploying Phones From The Provisioning Server

    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 phone firmware. The configuration files, supplied with each firmware release, must be used with that release.

  • Page 44: Dhcp

    Static: You can manually configure the server address via phone user interface or  web user interface. For more information on the above methods, refer to Yealink 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.
  • Page 45 Getting Started The following table lists common DHCP options supported by IP phones. Parameter DHCP Option Description Subnet Mask Specify the client’s subnet mask. Specify the offset of the client's subnet in Time Offset seconds from Coordinated Universal Time (UTC). Specify a list of IP addresses for routers on the Router client’s subnet.
  • Page 46 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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.
  • Page 47 Getting Started Parameters Permitted Values Default 1-PPPoE (not applicable to SIP-T42G/T41P IP phones) 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->Settings->Advanced Settings (default password: admin) ->Network->WAN Port->IPv4 network.static_dns_enable...
  • Page 48 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Menu->Settings->Advanced Settings (default password: admin) ->Network->WAN Port->IPv4->DHCP IPv4 Client->Static DNS (Enabled) ->IPv4 Pri.DNS network.secondary_dns IPv4 Address Blank 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...
  • Page 49 Getting Started In the IPv4 Config block, mark the DHCP radio box. Mark the Static DNS radio box. Enter the desired values in the Primary DNS and Secondary DNS fields. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone.
  • Page 50 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure Network parameters can be configured manually using the configuration files or locally. Configure network parameters of the IP phone manually. Parameters: network.internet_port.type network.ip_address_mode Configuration File <MAC>.cfg 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.
  • Page 51 Getting Started Parameters Permitted Values Default 0, 1 or 2 network.ip_address_mode 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->Settings->Advanced Settings (default password: admin) ->Network->WAN...
  • Page 52 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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.
  • Page 53 Getting Started 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...
  • Page 54 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select desired value from the pull-down list of Mode (IPv4/IPv6). Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone. To configure a static IPv4 address via web user interface: Click on Network->Basic.

  • Page 55: Pppoe

    Getting Started To configure the IP address mode via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->WAN Port. Press to select IPv4 or IPv4 & IPv6 from the IP Mode field. Press the Save soft key to accept the change. The IP phone reboots automatically to make settings effective after a period of time.
  • Page 56 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones http://<phoneIPAddress>/servlet ?p=network&q=load Phone User Interface Configure PPPoE on the IP phone. Details of Configuration Parameters: Parameters Permitted Values Default network.internet_port.type 0, 1 or 2 Description: Configures the Internet (WAN) port type for IPv4 when the IP address mode is configured as IPv4 or IPv4 &...
  • Page 57 Getting Started Parameters Permitted Values Default String within 99 network.pppoe.password Blank characters 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...
  • Page 58 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones To configure PPPoE via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->WAN Port->IPv4->PPPoE IPv4 Client. Enter the user name and password in corresponding fields. Press the Save soft key to accept the change. The IP phone reboots automatically to make settings effective after a period of time.
  • Page 59 Internet port and PC port for the IP phone to transmit in 10Mbps, 100Mbps or 1000Mbps (1000Mbps is only applicable to SIP-T48G/T46G/T42G/T29G/T23G IP phones). Procedure The transmission methods of Ethernet ports can be configured using the configuration files or locally.
  • Page 60 2-Full Duplex, 100Mbps 3-Half Duplex, 10Mbps 4-Half Duplex, 100Mbps 5-Full Duplex, 1000Mbps (only applicable to SIP-T48G/T46G/T42G/T29G/T23G IP phones) 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.
  • Page 61 Permitted Values Default 4-Half Duplex, 100Mbps 5-Full Duplex, 1000Mbps (only applicable to SIP-T48G/T46G/T42G/T29G/T23G IP phones) 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.
  • Page 62 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones files. Procedure PC port mode can be configured using the configuration files or locally. Configure the PC port. Configuration File <y0000000000xx>.cfg Parameter: network.pc_port.enable Configure the PC port mode. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=network-pcport&q=load Details of Configuration Parameters:...
  • Page 63 Automatically, from the provisioning server for a mass of phones.  The following table lists the associated and latest firmware name for each IP phone model (X is replaced by the actual firmware version). IP Phone Model Associated Firmware Name Firmware Name Example SIP-T48G 35.x.x.x.rom 35.80.0.20.rom SIP-T46G 28.x.x.x.rom 28.80.0.20.rom...

  • Page 64: Upgrading Firmware

    52.80.0.20.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 firmware to your local system in advance.
  • Page 65 Getting Started Upgrade Firmware from the Provisioning Server IP phones support using FTP , TFTP , HTTP and HTTPS protocols to download 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 configuration files and then download firmware during startup.
  • Page 66 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Details of Configuration Parameters: Parameters Permitted Values Default auto_provision.power_on 0 or 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...
  • Page 67 Getting Started Parameters Permitted Values Default auto_provision.weekly.enable 0 or 1 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...
  • Page 68 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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.
  • Page 69 Getting Started To configure the way for the IP phone to check for configuration files via web user interface: Click on Settings->Auto Provision. Make the desired change. Click Confirm to accept the change. When the “Power On” is set to On, the IP phone will check configuration files stored on the provisioning server during startup and then will download firmware from the server.
  • Page 70 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones...

  • Page 71: Configuring Basic Features

    Configuring Basic Features This chapter provides information for making configuration changes for the following basic features: Power Indicator LED  Notification Popups  Contrast  Wallpaper  Backlight  Call Display  Web Server Type  User Password  Administrator Password ...

  • Page 72: Power Indicator Led

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Anonymous Call Rejection  Do Not Disturb  Busy Tone Delay  Return Code When Refuse  Early Media  180 Ring Workaround  Use Outbound Proxy in Dialog  SIP Session Timer  Session Timer ...
  • Page 73 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.
  • Page 74 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Details of Configuration Parameters: Permitted Parameters Default Values phone_setting.common_power_led_enable 0 or 1 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) Web User Interface: Features->Power LED->Common Power Light On Phone User Interface:...
  • Page 75 Configuring Basic Features Permitted Parameters Default Values 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...

  • Page 76: Notification Popups

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired value from the pull-down list of Mute Power Light Flash. Select the desired value from the pull-down list of Hold/Held Power Light Flash. Select the desired value from the pull-down list of Talk/Dial Power Light On. Click Confirm to accept the change.
  • Page 77 Configuring Basic Features Details of Configuration Parameters: Permitted Parameters Default Values features.voice_mail_popup.enable 0 or 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).
  • Page 78 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Permitted Parameters Default Values Phone User Interface: None features.text_message_popup.enable 0 or 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 Web User Interface: Features->Notification Popups->Display Text Message Popup Phone User Interface:...

  • Page 79: Contrast

    You can configure the LCD’s contrast of SIP-T27P , SIP-T23P/G and SIP-T21(P) E2 IP phones, EXP39 connected to SIP-T29G/T27P IP phones and EXP40 connected to SIP-T48G/T46G IP phones. Make sure the expansion module has been connected to the IP phone before adjustment.

  • Page 80: Wallpaper

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default Note: We recommend that you set the contrast of the LCD screen to 6 as a more comfortable level. Web User Interface: Settings->Preference->Contrast Phone User Interface: Menu->Settings->Basic Settings->Display->Contrast To configure contrast via web user interface: Click on Settings->Preference.
  • Page 81 Configuring Basic Features The wallpaper image format must meet the following: Phone Model Format Resolution Single File Size Total File Size SIP-T48G .jpg/.png/.bmp <=800*480 <=5MB <=20MB SIP-T46G/T29G .jpg/.png/.bmp <=480*272 <=5MB <=20MB Procedure Wallpaper can be configured using the configuration files or locally.
  • Page 82 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 The default value is Default.png. Note: It is only applicable to SIP-T48G/T46G/T29G IP phones. Web User Interface: Settings->Preference->Wallpaper Phone User Interface: Menu->Basic->Display->Wallpaper...
  • Page 83 Configuring Basic Features Click Upload to upload the file. Click Confirm to accept the change. The custom wallpaper appears in the pull-down list of Wallpaper. To change the wallpaper via web user interface: Click on Settings->Preference. Select the desired wallpaper from the pull-down list of Wallpaper. Click Confirm to accept the change.

  • Page 84: Backlight

    Backlight time is applicable to SIP-T48G/T46G/T42G/T41P/T29G/T27P/T23P/T23G/T21(P) E2 IP phones and EXP40 connected to SIP-T48G/T46G IP phones and EXP39 conneted to SIP-T29G/T27P IP phones. You can configure the backlight time as one of the following types: Always Off: Backlight is turned off permanently (not applicable to ...
  • Page 85 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, SIP-T29G/T27P IP phones and the connected EXP39. Web User Interface: Settings->Preference->Backlight Active Level...
  • Page 86 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 0-Off 1-Low Note: It is only applicable to SIP-T48G/T46G/T29G IP phones. Web User Interface: Settings->Preference->Backlight Inactive Level Phone User Interface: Menu->Basic->Display->Backlight->Backlight Inactive Level Refer to 0, 1, 15, 30, 60, 120, 300, 600 phone_setting.backlight_time...
  • Page 87 Configuring Basic Features For SIP-T27P/T23P/T23G/T21(P) E2: The default value is 30. Web User Interface: Settings->Preference->Backlight Time (seconds) Phone User Interface: Menu->Settings->Basic Settings->Display->Backlight->Backlight Time To configure backlight via web user interface (take SIP-T23G IP phones for example): Click on Settings->Preference. Select the desired value from the pull-down list of Backlight Time (seconds). Click Confirm to accept the change.

  • Page 88: Call Display

    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/T46G/T29G IP phones. Display called party information allows the IP phone to present the callee identity in...
  • Page 89 Configuring Basic Features The following figure shows an example of screen display when Display Called Party Information feature is enabled on the phone. The following shows an incoming call from 1008 to 1009. 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:xxx@domain.com>).
  • Page 90 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/T29G IP phones. Web User Interface: Settings->Call Display->Display Contact Photo Phone User Interface: None phone_setting.called_party_info_display.enable...

  • Page 91: Web Server Type

    Configuring Basic Features Parameters Permitted Values Default Settings->Call Display->Call Information Display Method Phone User Interface: None To configure call display feature via web user interface: Click on Settings->Call Display. Select the desired value from the pull-down list of Display Called Party Information. The default value is Disabled.
  • Page 92 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Configure the web access type, HTTP port and HTTPS port. Web User Interface Navigate to: Local http://<phoneIPAddress>/servl et?p=network-adv&q=load Configure the web access type, Phone User Interface HTTP port and HTTPS port. Details of Configuration Parameters: Parameters Permitted Values Default...
  • Page 93 Configuring Basic Features Parameters Permitted Values Default wui.https_enable 0 or 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.

  • Page 94: User Password

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones The default HTTPS port number is 443. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone. To configure web server type via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->Webserver Type.
  • Page 95 Configuring Basic Features password as soon as possible. Procedure User password can be changed using the configuration files or locally. Change the user password of the IP phone. Configuration File <y0000000000xx>.cfg Parameter: security.user_password Change the user password of the IP phone. Local Web User Interface Navigate to:...

  • Page 96: Administrator Password

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Valid characters are ASCII characters 32-126(0x20-0x7E) except 58(3A). Click Confirm to accept the change. Note If logging into the web user interface of the phone with the user credential, you need to enter the old user password in the Old Password field. Advanced menu options are strictly used by administrators.
  • Page 97 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:123 means setting the password of administrator (current user name is “admin”) to password 123.

  • Page 98: Phone Lock

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Valid characters are ASCII characters 32-126(0x20-0x7E). Press the Save soft key to accept the change. Phone lock is used to lock the IP phone to prevent it from unauthorized use. Once the IP phone is locked, a user must enter the password to unlock it. IP phones offer three types of phone lock: Menu Key, Function Keys and All Keys.
  • Page 99 2-Menu Keys For more information, refer to Phone Lock Type on page 86. Note: It is not applicable to SIP-T48G IP phones. It works only if the parameter “phone_setting.phone_lock.enable” is set to 1(Enabled). Web User Interface: Features->Phone Lock->Phone Lock Type Phone User Interface: Menu->Settings->Advanced Settings (default password: admin) ->Phone...
  • Page 100 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Lock->Lock Type phone_setting.phone_lock.unlock_pin characters within 15 digits Description: Configures the password for unlocking the phone. Web User Interface: Features->Phone Lock->Phone Unlock PIN (0~15 Digit) Phone User Interface: Menu->Settings->Basic Settings->Change PIN phone_setting.phone_lock.lock_time_out Integer from 0 to 3600 Description:...
  • Page 101 Refer to the programablekey.X.type following content 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 29 (for SIP-T48G)
  • Page 102 X ranges from 1 to 21 (for SIP-T27P) X ranges from 1 to 3 (for SIP-T23P/G) X ranges from 1 to 2 (for SIP-T21(P) E2) For programable keys: X=1-10, 12-14 (for SIP-T48G/T46G) X=1-10, 13 (for SIP-T42G/T41P) X=1-14 (for SIP-T29G/T27P) X=1-10, 14 (for SIP-T23P/T23G/T21(P) E2) Example: linekey.1.type = 50...
  • Page 103 Configuring Basic Features Parameter Permitted Values Default 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).
  • Page 104 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default When X=13, the default value is 0 (NA). When X=14, the default value is 2 (Forward). For SIP-T23P/T23G/T21(P) E2 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).
  • Page 105 Configuring Basic Features Enter the desired time in the Phone Lock Time Out (0~3600s) field. Click Confirm to accept the change. To configure a phone lock key via web user interface: Click on DSSKey->Line Key (Programable Key). In the desired DSS key field, select Phone Lock from the pull-down list of Type. (Optional.) Enter the string that will appear on the LCD screen in the Label field.

  • Page 106: Time And Date

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Enter the new unlock PIN again in the Confirm PIN field. Press the Save soft key to accept the change. To configure a phone lock key via phone user interface: Press Menu->Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Phone Lock from the Type field.
  • Page 107 Configuring Basic Features Option Configuration Methods Configuration Files Time Format Web User Interface Phone User Interface Web User Interface Date Phone User Interface Configuration Files Date Format Web User Interface Phone User Interface Configuration Files Daylight Saving Time Web User Interface Procedure Configuration changes can be performed using the configuration files or locally.
  • Page 108 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones formats. Parameters: local_time.time_format local_time.date_format Configure NTP by DHCP priority feature. Configure the NTP server, time zone and DST. Configure the time and date Web User Interface manually. Configure the time and date formats. Local Navigate to: http://<phoneIPAddress>/servlet ?p=settings-datetime&q=load...
  • Page 109 Configuring Basic Features Parameters Permitted Values Default 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 &...
  • Page 110 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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 local_time.time_zone -11 to +14 Description:...
  • Page 111 Configuring Basic Features Parameters Permitted Values Default local_time.summer_time 0, 1 or 2 Description: Configures Daylight Saving Time (DST) feature. 0-Disabled 1-Enabled 2-Automatic Web User Interface: Settings->Time & Date->Daylight Saving Time Phone User Interface: Menu->Settings->Basic Settingss->Time & Date->SNTP Settings->Daylight Saving local_time.dst_time_type 0 or 1 Description: Configures the DST time type.
  • Page 112 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Month: 1=January, 2=February,…, 12=December Week of Month: 1=the first week in a month,…, 5=the last week in a month Day of Week: 1=Monday, 2=Tuesday,…, 7=Sunday Hour of Day: 0=0am, 1=1am,…, 23=11pm Note: It works only if the parameter “local_time.summer_time”...
  • Page 113 Configuring Basic Features Parameters Permitted Values Default Note: It works only if 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 Description: Configures the IP phone to obtain time from the NTP server or manual settings. 0-NTP 1-Manual Web User Interface:...
  • Page 114 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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 Note: “WWW” represents the abbreviation of the week, “DD” represents a two-digit day, “MMM” represents the first three letters of the month, “YYYY” represents a four-digit year, and “YY”...
  • Page 115 Configuring Basic Features Select the desired time zone from the pull-down list of Time Zone. Enter the domain names or IP addresses in the Primary Server and Secondary Server fields respectively. Enter the desired time interval in the Synchronism (15~86400s) field. Select the desired value from the pull-down list of Daylight Saving Time.
  • Page 116 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired values of DST Start Month, DST Start Week of Month, DST Start Day of Week, Start Hour of Day; DST Stop Month, DST Stop Week of Month, DST Stop Day of Week and End Hour of Day from the pull-down lists. Enter the desired offset time in the Offset (minutes) field.
  • Page 117 Configuring Basic Features Select the desired value from the pull-down list of Date Format. Click Confirm to accept the change. To configure the NTP server and time zone via phone user interface: Press Menu->Settings->Basic Settings->Time & Date->SNTP Settings. Press , or the Switch soft key to select the time zone that applies to your area from the Time Zone field.

  • Page 118: Language

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Press , or the Switch soft key to select the desired date format from the Date Format field. Press the Save soft key to accept the change. IP phones support multiple languages. Languages used on the phone user interface and web user interface can be specified respectively as required.
  • Page 119 Configuring Basic Features Available Language Associated Language Pack German 004.GUI.German.lang Italian 005.GUI.Italian.lang Polish 006.GUI.Polish.lang Portuguese 007.GUI.Portuguese.lang Spanish 008.GUI.Spanish.lang Turkish 009.GUI.Turkish.lang Russian 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).
  • Page 120 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones The following shows a portion of the language pack “000.GUI.English.lang” for the phone user interface (take SIP-T23G IP phones for example): The following table lists available languages and associated language packs for the web user interface: Associated Note Available Language Associated Language Pack...
  • Page 121 Configuring Basic Features Associated Note Available Language Associated Language Pack Language Pack Russian 11.Russian.js 11.Russian_note.xml 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).
  • Page 122 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Don't modify the name of the note field. The following shows a portion of the note language pack “1.English_note.xml” for the web user interface (take SIP-T23G IP phones for example): Note The new added language must be supported by the font library on the IP phone. If the characters in the custom language file are not supported by the phone, the IP phone will display “?”...
  • Page 123 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”...
  • Page 124 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default 001.GUI.Chinese_S.lang) gui_lang.delete = http://localhost/001.GUI.Chinese_S.lang Web User Interface: None Phone User Interface: None wui_lang.url URL within 511 characters 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”...
  • Page 125 Configuring Basic Features Parameter Permitted Values Default Web User Interface: None Phone User Interface: None http://localhost/all or wui_lang.delete Blank Y.name.js http://localhost/ Description: Delete all customized language packs and note language packs of the web user interface. Example: Delete all customized language packs: wui_lang.delete = http://localhost/all Delete a customized language pack (e.g., 11.Russian.js) of the web user interface.
  • Page 126 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones lang.wui Specify the language for the web user interface. Web User Interface Navigate to: http://<phoneIPAddress>/servlet Local ?p=settings-preference&q=load Specify the language for the Phone User Interface phone user interface. Details of Configuration Parameters: Parameters Permitted Values Default lang.gui Refer to the following content...

  • Page 127: Input Method Customization

    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” or “Russian_ime.txt”, and then download it to the IP phone. If you choose Russian language for the phone, the input method file will be “Russian_ime.txt”.
  • Page 128 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones The following shows a portion of the input method file “ime.txt”:...
  • Page 129 Configuring Basic Features The following shows a portion of the input method file “Russian_ime.txt”:...
  • Page 130 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones You can add new characters or adjust the character order of the existing input method. The following show an example of adding some German 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.
  • Page 131 Configuring Basic Features when editing contacts. Parameter: directory.edit_default_input_meth 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 Blank URL within 511 characters 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.

  • Page 132: Logo Customization

    Logo customization allows unifying the IP phone appearance or displaying a custom image on the idle screen such as a company logo, instead of the default system logo. Logo is not applicable to SIP-T48G, SIP-T46G and SIP-T29G IP phones. These three IP phone models use wallpaper instead.
  • Page 133 <=132*64 2 gray scale Note Before uploading your custom logo to IP phones, ensure your logo file is correctly formatted. For more information on customizing a logo file, refer to Yealink IP Phones Auto Provisioning Guide. Procedure The logo shown on the idle screen can be configured using the configuration files or locally.
  • Page 134 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/T29G IP phones. Web User Interface: Features->General Information->Use Logo...

  • Page 135: Softkey Layout

    Configuring Basic Features Select Custom logo from the pull-down list of Use Logo. Click Browse to select the logo file from your local system. Click Upload to upload the file. Click Confirm to accept the change. The image 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’...
  • Page 136 End Call Connecting Transfer Empty Empty Switch SemiAttendTrans Empty End Call Send Empty History Delete Switch End Call Line Dialing (not applicable to SIP-T48G) Favorite (Directory) GPickup DPickup Retrieve Empty Empty Empty Switch RingBack Empty End Call RingBack Transfer Empty Empty...
  • Page 137 Empty Empty Switch Held Empty Answer End Call Reject NewCall Transfer Empty PreTrans (not Directory applicable to Delete Switch SIP-T48G) End Call Send Empty Empty Hold Switch Split Answer Conferenced End Call Reject Mute Manager RTP Status Procedure Softkey layout can be configured using the configuration files or locally.
  • Page 138 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones custom_softkey_ring_back.url custom_softkey_talking.url Configure the softkey layout. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=settings-softkey&q=load Details of Configuration Parameters: Parameters Permitted Values Default phone_setting.custom_softkey_enable 0 or 1 Description: Enables or disables custom soft keys layout feature. 0-Disabled 1-Enabled Web User Interface:...
  • Page 139 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”...
  • Page 140 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Phone User Interface: None custom_softkey_ring_back.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 RingBack state. Example: The following example uses HTTP to download the RingBack state file from the “XMLfiles”...
  • Page 141 Configuring Basic Features soft keys are selected, a More soft key will appear on the LCD screen, and the selected soft keys are displayed in two pages. Repeat the step 4 to add more soft keys to the Selected Softkeys column. To remove the soft key from the Selected Softkeys column, select the desired soft key and then click To adjust the display order of soft keys, select the desired soft key and then click...
  • Page 142 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter: features.key_tone Configure a send key. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Web User Interface Configure a send sound and key tone. Local Navigate to: http://<phoneIPAddress>/servlet ?p=features-audio&q=load Configure the send key. Phone User Interface Configure a key tone. Details of Configuration Parameters: Parameters Permitted Values...
  • Page 143 Configuring Basic Features Parameters Permitted Values Default 1-Enabled If it is set to 1 (Enabled), the IP phone will play a tone when a user presses a key on your phone keypad. Web User Interface: Features->Audio->Key Tone Phone User Interface: Menu->Settings->Basic Settings->Sound->Key Tone features.send_key_tone 0 or 1...
  • Page 144 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired value from the pull-down list of Key As Send. Click Confirm to accept the change. To configure a send sound and key tone via web user interface: Click on Features->Audio. Select the desired value from the pull-down list of Key Tone. Select the desired value from the pull-down list of Send Sound.

  • Page 145: Dial Plan

    Configuring Basic Features Press , or the Switch soft key to select the desired type from the Key Tone field. 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.

  • Page 146: Replace Rule

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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.
  • Page 147 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. Permitted Values: 0 to 16 (for SIP-T48G/T46G/T29G) 0 to 12 (for SIP-T42G) 0 to 6 (for SIP-T41P/T27P)
  • Page 148 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Note: Multiple line IDs are separated by commas. Web User Interface: Settings->Dial Plan->Replace Rule->Account Phone User Interface: None dialplan_replace_rule.url URL within 511 characters 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:...

  • Page 149: Dial-Now

    Configuring Basic Features If you leave this field blank or enter 0, the replace rule will apply to all accounts on the IP phone. Click Add to add the replace rule. Dial-now is a string used to match numbers entered by the user. When entered numbers match the predefined dial-now rule, the IP phone will automatically dial out the numbers without pressing the send key.
  • Page 150 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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 Local Web User Interface Configure the delay time for the dial-now rule. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Details of Configuration Parameters: Parameters...
  • Page 151 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. Permitted Values: 0 to 16 (for SIP-T48G/T46G/T29G) 0 to 12 (for SIP-T42G) 0 to 6 (for SIP-T41P/T27P)
  • Page 152 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default None To create a dial-now rule via web user interface: Click on Settings->Dial Plan->Dial-now. Enter the desired value in the Rule field. Enter the desired line ID in the Account field or leave it blank. If you leave this field blank or enter 0, the dial-now rule will apply to all accounts on the IP phone.

  • Page 153: Area Code

    Configuring Basic Features Enter the desired time within 1-14 (in seconds) in the Time-Out for Dial-Now Rule field. Click Confirm to accept the change. Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country. When entered numbers match the predefined area code rule, the IP phone will automatically add the area code before the numbers when dialing out them.
  • Page 154 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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”...
  • Page 155 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. Permitted Values: 0 to 16 (for SIP-T48G/T46G/T29G) 0 to 12 (for SIP-T42G) 0 to 6 (for SIP-T41P/T27P)

  • Page 156: Block Out

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Click Confirm to accept the change. Block out rule prevents users from dialing out specific numbers. When entered numbers match the predefined block out rule, the LCD screen prompts “Forbidden Number”. IP phones support up to 10 block out rules. Procedure Block out rule can be created using the configuration files or locally.
  • Page 157 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 Permitted Values: 0 to 16 (for SIP-T48G/T46G/T29G) 0 to 12 (for SIP-T42G) 0 to 6 (for SIP-T41P/T27P)

  • Page 158: Hotline

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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.
  • Page 159 Configuring Basic Features Parameter Permitted Values Default hotline feature. Example: features.hotline_number = 1234 Web User Interface: Features->General Information->Hotline Number Phone User Interface: Menu->Features->Hot Line->Hot Number features.hotline_delay Integer from 0 to 10 Description: Configures the waiting time (in seconds) for the IP phone to automatically dial out the hotline number.

  • Page 160: Off Hook Hot Line Dialing

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Enter the delay time in the Hotline Delay (0~10s) field. Click Confirm to accept the change. To configure hotline via phone user interface: Press Menu->Features->Hot Line. Enter the hotline number in the Hot Number field. Enter the waiting time (in seconds) in the Hotline Delay field.
  • Page 161 X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).

  • Page 162: Directory

    Note: It works only if the value of the parameter “account.X.auto_dial_enable” is set to 1 (Enabled). X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 163 Configuring Basic Features Details of the Configuration Parameter: Parameter Permitted Values Default directory_setting.url URL within 511 characters Blank 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: Click on Directory->Setting.
  • Page 164 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones LDAP . The search source list can be configured using a super search file. For more information on how to customize a super search template, refer to Super Search Template on page 516. Procedure Search source list in dialing can be configured using the configuration files or locally.

  • Page 165: Call Log

    Configuring Basic Features The LCD screen displays the search results in the adjusted order. Click Confirm to accept the change. Call log contains call information such as remote party identification, time and date, and call duration. It can be used to redial previous outgoing calls, return incoming calls, and save contact information from call log lists to the contact directory.
  • Page 166 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Details of the Configuration Parameter: Parameter Permitted Values Default features.save_call_history 0 or 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 missed calls, placed calls, received calls and the forwarded calls in the call log lists.

  • Page 167: Missed Call Log

    If it is set to 1 (Enabled), a prompt message "<number> New Missed Call(s)" along with an indicator icon is displayed on the IP phone idle screen when the IP phone misses calls. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G).

  • Page 168: Local Directory

    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 page 518.
  • Page 169 Configuring Basic Features Procedure Configuration changes can be performed using the configuration files or locally. Specify the access URL of the local contact file (*.xml). Configuration File <y0000000000xx>.cfg Parameter: local_contact.data.url Add a new group and a contact to the local directory. To import or export the local contact file.
  • Page 170 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired ring tone from the pull-down list of Ring. Click Add to add the group. To add a contact to the local directory via web user interface: Click on Directory->Local Directory. In the Directory block, enter the name and the office, mobile or other numbers in the corresponding fields.
  • Page 171 Configuring Basic Features If Auto is selected, the IP phone will use the first available account when placing calls to the contact from the local directory. Click Add to add the contact. To add a group to the local directory via phone user interface: Press Menu->Directory->Local Directory.
  • Page 172 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Click Browse to locate a contact list file (the file format must be *.xml) from your local system. Click Import XML to import the contact list. The web user interface prompts "The original contact will be covered, Continue?". Click OK to complete importing the contact list.

  • Page 173: Live Dialpad

    Configuring Basic Features At least one item should be selected to be imported into the local directory. Click Import to complete importing the contact list. To export a contact list via web user interface: Click on Directory->Local Directory. Click Export XML (or Export CSV). Click Save to save the contact list to your local system.
  • Page 174 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure Live dialpad can be configured using the configuration files or locally. Configure live dialpad. Parameters: Configuration File <y0000000000xx>.cfg phone_setting.predial_autodial phone_setting.inter_digit_time Configure live dialpad. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=settings-preference&q=load Details of Configuration Parameters: Parameters Permitted Values Default...

  • Page 175: Call Waiting

    Configuring Basic Features To configure live dialpad via web user interface: Click on Settings->Preference. Select the desired value from the pull-down list of Live Dialpad. Enter the desired delay time in the Inter Digit Time (1~14s) field. Click Confirm to accept the change. Call waiting allows IP phones to receive a new incoming call when there is already an active call.
  • Page 176 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Configure call waiting tone. Navigate to: http://<phoneIPAddress>/servlet ?p=features-audio&q=load Configure call waiting and call Phone User Interface waiting tone. Details of Configuration Parameters: Parameters Permitted Values Default call_waiting.enable 0 or 1 Description: Enables or disables call waiting feature.
  • Page 177 Configuring Basic Features Parameters Permitted Values Default Features->Audio->Call Waiting Tone Phone User Interface: Menu->Features->Call Waiting->Play Tone 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.
  • Page 178 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones (Optional.) Enter the call waiting off code in the Call Waiting Off Code field. Click Confirm to accept the change. To configure call waiting tone via web user interface: Click on Features->Audio. Select the desired value from the pull-down list of Call Waiting Tone. Click Confirm to accept the change.

  • Page 179: Auto Redial

    Configuring Basic Features (Optional.) Enter the call waiting off code in the Off Code field. 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.
  • Page 180 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default auto_redial.interval Integer from 1 to 300 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.

  • Page 181: Auto Answer

    Configuring Basic Features The default value is 10. Click Confirm to accept the change. To configure auto redial via phone user interface: Press Menu->Features->Auto Redial. Press , or the Switch soft key to select the desired value from the Auto Redial field.
  • Page 182 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 183 Configuring Basic Features Parameters Permitted Values Default For SIP-T48G: Menu->Features->Auto Answer->Line X->On/Off features.auto_answer_delay Integer from 1 to 4 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)
  • Page 184 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Enter the desired time in the Auto-Answer Delay (1~4s) field. Click Confirm to accept the change. To configure auto answer via phone user interface (take T23G IP phones for example): Press Menu->Features->Auto Answer. Press , or the Switch soft key to select the desired value from the Line ID field.

  • Page 185: Call Completion

    Configuring Basic Features 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 ...

  • Page 186: Anonymous Call

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default Menu->Features->Call Completion->Call Completion To configure call completion via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Call Completion. Click Confirm to accept the change. To configure call completion via phone user interface: Press Menu->Features->Call Completion.
  • Page 187 Contact: <sip:1009@10.3.20.14:5060> Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T23G 44.80.0.20 Allow-Events: talk,hold,conference,refer,check-sync P-Preferred-Identity: <sip:1009@10.3.5.199> Privacy: id 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.
  • Page 188 The callee’s phone LCD screen presents anonymous instead of the caller’s identity. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 189 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 190 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default account.1.anonymous_call_offcode = *73 Note: It works only if 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 Call->Off Code To configure anonymous call via web user interface: Click on Account.

  • Page 191: Anonymous Call Rejection

    Configuring Basic Features from the Send Anony Code field. (Optional.) Enter the anonymous call on code in the On Code field. (Optional.) Enter the anonymous call off code in the Off Code field. Anonymous call rejection allows IP phones to automatically reject incoming calls from callers whose identity has been deliberately concealed.
  • Page 192 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 193 Configures the anonymous call rejection on code to activate the server-side anonymous call rejection feature for account X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).

  • Page 194: Do Not Disturb

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired value from the pull-down list of Local Anonymous Rejection. Select the desired value from the pull-down list of Send Anonymous Rejection code. (Optional.) Enter the Send Anonymous Rejection on code in the On Code field. (Optional.) Enter the Send Anonymous Rejection off code in the Off Code field.
  • Page 195 Configuring Basic Features A user can activate or deactivate DND using the DND key or DND soft key. 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.
  • Page 196 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 197 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/T29G). X ranges from 1 to 12 (for SIP-T42G).
  • Page 198 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default features.dnd_mode 0 or 1 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.
  • Page 199 Configuring Basic Features Parameters Permitted Values Default 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.
  • Page 200 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 201 Permitted Values Default The default value is 15. 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).
  • Page 202 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default 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=11, the default value is 0 (NA).
  • Page 203 Configuring Basic Features Click Confirm to accept the change. To configure DND feature via web user interface: Click on Features->Forward & DND. 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.
  • Page 204 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 4) (Optional.) Enter the DND off code in the DND Off Code field. Click Confirm to accept the change. To specify the return code and the reason when DND is enabled via web user interface: Click on Features->General Information.

  • Page 205: Busy Tone Delay

    Configuring Basic Features Click Confirm to accept the change. To configure a DND key via phone user interface: Press Menu->Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field. Press , or the Switch soft key to select DND from the Key Type field.
  • Page 206 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Details of the Configuration Parameter: Parameter Permitted Values Default features.busy_tone_delay 0, 3 or 5 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.

  • Page 207: Return Code When Refuse

    Configuring Basic Features Return code when refuse defines the return code and reason of the SIP response message for the refused call. The caller’s phone LCD screen displays the reason according to the received return code. Available return codes and reasons are: 404 (Not found) ...

  • Page 208: Early Media

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default Features->General Information->Return Code When Refuse Phone User Interface: None To specify the return code and the reason when refusing a call via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Return Code When Refuse. Click Confirm to accept the change.
  • Page 209 Configuring Basic Features Procedure 180 ring workaround can be configured using the configuration files or locally. Configure 180 ring workaround. Configuration File <y0000000000xx>.cfg Parameter: phone_setting.is_deal180 Configur 180 ring workaround. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=features-general&q=load Details of the Configuration Parameter: Parameter Permitted Values Default...

  • Page 210: Use Outbound Proxy In Dialog

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired value from the pull-down list of 180 Ring Workaround. 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.
  • Page 211 Configuring Basic Features Details of the Configuration Parameter: Parameter Permitted Values Default sip.use_out_bound_in_dialog 0 or 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.

  • Page 212: Sip Session Timer

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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.
  • Page 213 Configuring Basic Features Parameters Permitted Values Default Description: Configures the session timer T2 (in seconds). T2 represents the maximum retransmit interval for non-INVITE requests and INVITE responses. Web User Interface: Settings->SIP->SIP Session Timer T2 (2~40s) Phone User Interface: None sip.timer_t4 Float from 2.5 to 60 Description: Configures the session timer of T4 (in seconds).

  • Page 214: Session Timer

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones The default value is 5s. Click Confirm to accept the change. 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.
  • Page 215 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 216 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/T29G). X ranges from 1 to 12 (for SIP-T42G).

  • Page 217: Call Hold

    Configuring Basic Features Select the desired refresher from the pull-down list of Session Refresher. 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.
  • Page 218 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure Call hold can be configured using the configuration files or locally. Configure the call hold tone and call hold tone delay. Parameters: features.play_hold_tone.enable features.play_hold_tone.delay <y0000000000xx>.cfg Specify whether RFC 2543 (c=0.0.0.0) outgoing hold Configuration File signaling is used.
  • Page 219 Configuring Basic Features 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 features.play_hold_tone.delay...
  • Page 220 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:moh@sip.com, <sip:moh@sip.com>, <yealink.com> or yealink.com. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 221 Configuring Basic Features Select the desired value from the pull-down list of RFC 2543 Hold. Click Confirm to accept the change. To configure call hold tone and call hold tone delay via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Play Hold Tone. Enter the desired time in the Play Hold Tone Delay field.

  • Page 222: Call Forward

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones To configure MoH via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced. Enter the SIP URI (e.g., sip:moh@sip.com) in the Music Server URI field. Click Confirm to accept the change.
  • Page 223 Configuring Basic Features 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.
  • Page 224 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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 Configure diversion/history-info feature. Parameter: features.fwd_diversion_enable Configure forward international. Parameter: forward.international.enable Configure call forward. Navigate to: http://<phoneIPAddress>/servlet?p =features-forward&q=load Web User Configure diversion/history-info Interface feature.
  • Page 225 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 226 X on the IP phone. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 227 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/T29G). X ranges from 1 to 12 (for SIP-T42G).
  • Page 228 X on the IP phone. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 229 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/T29G). X ranges from 1 to 12 (for SIP-T42G).
  • Page 230 Custom. Incoming calls will be forwarded when not answered after N*6 seconds. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 231 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. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G).
  • Page 232 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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...
  • Page 233 Configuring Basic Features Parameters Permitted Values Default 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:...
  • Page 234 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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.
  • Page 235 Configuring Basic Features Parameters Permitted Values Default 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...
  • Page 236 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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.
  • Page 237 Configuring Basic Features To configure call forward via web user interface: Click on Features->Forward & DND. In the Forward block, mark the desired radio box in the Mode field. a) If you mark the Phone radio box: 1) Mark the desired radio box in the Always/Busy/No Answer Forward field. 2) Enter the destination number you want to forward in the Target field.
  • Page 238 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 5) Select the ring time to wait before forwarding from the pull-down list of After Ring Time (0~120s) (only for the no answer forward). Click Confirm to accept the change. To configure Diversion/History-Info feature via web user interface: Click on Features->General Information.
  • Page 239 Configuring Basic Features Select the desired value from the pull-down list of Fwd International. Click Confirm to accept the change. To configure call forward in phone mode via phone user interface: Press Menu->Features->Call Forward. Press to select the desired forwarding type, and then press the Enter soft key.
  • Page 240 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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 the Switch soft key to select the ring time to wait before forwarding from the After Ring Time field.

  • Page 241: Call Transfer

    Configuring Basic Features 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 the Switch soft key to select the desired value from the No Answer Forward field.
  • Page 242 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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 feature determines whether to display the prompt "n New Missed Call(s)"...
  • Page 243 Configuring Basic Features Parameters Permitted Values Default None transfer.on_hook_trans_enable 0 or 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->Attend Transfer On Hook Phone User Interface: None...

  • Page 244: Network Conference

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired values from the pull-down lists of Semi-Attend Transfer, Blind Transfer on Hook and Attend Transfer on Hook. 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).
  • Page 245 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/T29G). X ranges from 1 to 12 (for SIP-T42G).

  • Page 246: Feature Key Synchronization

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired account from the pull-down list of Account. Click on Advanced. Select Network Conference from the pull-down list of Conference Type. Enter the conference URI in the Conference URI field. 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)
  • Page 247 Configuring Basic Features 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: Click on Features->General Information.

  • Page 248: Transfer On Conference Hang Up

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones For a conference call, 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.

  • Page 249: Directed Call Pickup

    Configuring Basic Features To configure Transfer on Conference Hang up via web user interface: Click on Features->Transfer. Select the desired value from the pull-down list of Transfer on Conference Hang up. 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.
  • Page 250 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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&q=load&model= Configure directed call pickup feature on a phone basis. Navigate to: Web User Interface http://<phoneIPAddress>/servl Local...
  • Page 251 Default Description: Configures the directed call pickup code for account X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 252 Configures a DSS key as a directed call pickup key on the IP phone. The digit 9 stands for the key type Direct Pickup. For line keys: X ranges from 1 to 29 (for SIP-T48G) X ranges from 1 to 27 (for SIP-T46G/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 253 For SIP-T23P/T23G/T21(P) E2 IP phones: The default value is 15. 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).
  • Page 254 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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). For SIP-T29G/T27P IP phones: When X=1, the default value is 28 (History).
  • Page 255 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 256 Configures the directed call pickup feature code followed by the monitored extension. For line keys: X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 257 Configuring Basic Features Click Confirm to accept the change. To configure directed call pickup feature on a phone basis via web user interface: Click on Features->Call Pickup. Select the desired value from the pull-down list of Directed Call Pickup. Enter the directed call pickup code in the Directed Call Pickup Code field. Click Confirm to accept the change.

  • Page 258: Group Call Pickup

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones To configure a directed pickup key via phone user interface: Press Menu->Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field. Press , or the Switch soft key to select DPickup from the Key Type field.
  • Page 259 Blank Description: Configures the group pickup code for account X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 260 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 261 Configuring Basic Features Parameters Permitted Values Default X ranges from 1 to 2 (for SIP-T21(P) E2) For programable keys: X=1-10, 12-14 (for SIP-T48G/T46G) X=1-10, 13 (for SIP-T42G/T41P) X=1-14 (for SIP-T29G/T27P) X=1-10, 14 (for SIP-T23P/T23G/T21(P) E2) Example: linekey.1.type = 23 Default:...
  • Page 262 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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).
  • Page 263 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 264 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 265 Configuring Basic Features To configure a group call pickup key via web user interface: Click on DSSKey->Line Key (or Programable Key). In the desired DSS key field, select Group Pickup from the pull-down list of Type. Enter the group call pickup code in the Value field. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
  • Page 266 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Enter the group call pickup code in the Group Call Pickup Code field. Click Confirm to accept the change. To configure a group pickup key via phone user interface: Press Menu->Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field.

  • Page 267: Dialog Info Call Pickup

    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/T29G). X ranges from 1 to 12 (for SIP-T42G).

  • Page 268: Recall

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G). X ranges from 1 to 2 (for SIP-T21(P) E2). Web User Interface: Account->Advanced->Dialog Info Call Pickup Phone User Interface: None To configure dialog info call pickup via web user interface:...
  • Page 269 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 270 Parameter Permitted Values Default 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/T29G 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.
  • Page 271 Configuring Basic Features Parameter Permitted Values Default 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).

  • Page 272: Call Park

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default When X=14, the default value is 2 (Forward). Web User Interface: DSSKey->Line Key/ Programable Key->Type Phone User Interface: Menu->Features->DSS Keys->Line Key X->Type To configure a recall key via web user interface: Click on DSSKey->Line Key (or Programable Key).
  • Page 273 Configures a DSS key as 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 274 1-16 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 275 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).

  • Page 276: Calling Line Identification Presentation

    If the caller already exists in the local directory, the local contact name assigned to the caller should be preferentially displayed and stored in the call log. Calling and For more information on calling line identification presentation, refer to 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.
  • Page 277 3-RPID-PAI-FROM 4-PAI-RPID-FROM 5-RPID-FROM X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G). X ranges from 1 to 2 (for SIP-T21(P) E2).

  • Page 278: Connected Line Identification Presentation

    If the callee already exists in the local directory, the local contact name assigned to the callee should be preferentially displayed. Calling and For more information on connected line identification presentation, refer to Connected Line Identification Presentation on Yealink IP Phones , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Procedure COLP can be configured only using the configuration files.

  • Page 279: Dtmf

    UPDATE message from the callee, and displays the identity in the From header. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 280 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones DTMF Keypad Frequencies: 1209 Hz 1336 Hz 1447 Hz 1633 Hz 697 Hz 770 Hz 852 Hz 941 Hz Three methods of transmitting DTMF digits on SIP calls: RFC 2833 -- DTMF digits are transmitted by RTP Events compliant to RFC 2833. ...
  • Page 281 Configuring Basic Features Procedure Configuration changes can be performed using the configuration files or locally. Configure the method of transmitting DTMF digit and the payload type. Parameters: <MAC>.cfg account.X.dtmf.type account.X.dtmf.dtmf_payload account.X.dtmf.info_type Configure the number of times Configuration File for the IP phone to send the end RTP Event packet.
  • Page 282 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/T29G). X ranges from 1 to 12 (for SIP-T42G).
  • Page 283 0-Disabled 1-DTMF-Relay 2-DTMF 3-Telephone-Event X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G). X ranges from 1 to 2 (for SIP-T21(P) E2).
  • Page 284 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Web User Interface: None Phone User Interface: None To configure the method of transmitting DTMF digits via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced.

  • Page 285: Suppress Dtmf Display

    Configuring Basic Features Select the desired value (1-3) from the pull-down list of DTMF Repetition. Click Confirm to accept the change. Suppress DTMF display allows IP phones to suppress the display of DTMF digits. DTMF digits are displayed as “*” on the LCD screen. Suppress DTMF display delay defines whether to display the DTMF digits for a short period of time before displaying as “*”.
  • Page 286 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Details of Configuration Parameters: Parameters Permitted Values Default features.dtmf.hide 0 or 1 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:...

  • Page 287: Transfer Via Dtmf

    Configuring Basic Features Select the desired value from the pull-down list of Suppress DTMF Display Delay. Click Confirm to accept the change. Call transfer is implemented via DTMF on some traditional servers. The IP phone sends specified DTMF digits to the server for transferring calls to third parties. Procedure Configuration changes can be performed using the configuration files or locally.
  • Page 288 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Details of Configuration Parameters: Parameters Permitted Values Default features.dtmf.replace_tran 0 or 1 Description: Enables or disables the IP phone to send DTMF sequences for transfer function when pressing the Transfer/Tran soft key or TRAN/TRANSFER key. 0-Disabled 1-Enabled If it is set to 0 (Disabled), the IP phone will perform the transfer as normal when...

  • Page 289: Intercom

    Configuring Basic Features Enter the specified DTMF digits in the Tran Send DTMF field. Click Confirm to accept the change. Intercom allows establishing an audio conversation directly. The IP phone can answer intercom calls automatically. This feature depends on support from a SIP server. Intercom is a useful feature in office environments to quickly connect with an operator or secretary.

  • Page 290: Intercom Key

    Configures a DSS key as 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 291 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 292 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).

  • Page 293: Incoming Intercom Calls

    Configuring Basic Features Select the desired line from the Account ID field. (Optional.) Enter the string that will appear on the LCD screen in the Label field. Enter the remote extension number in the Value field. Press the Save soft key to accept the change. The IP phone can process incoming calls differently depending on settings.
  • Page 294 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones feature. Details of Configuration Parameters: Parameters Permitted Values Default features.intercom.allow 0 or 1 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.
  • Page 295 Configuring Basic Features 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 parameter “features.intercom.allow” is set to 1 (Enabled). Web User Interface: Features->Intercom->Intercom Tone Phone User Interface: Menu->Features->Intercom->Intercom Tone...
  • Page 296 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired values from the pull-down lists of Accept Intercom, Intercom Mute, Intercom Tone and Intercom Barge. Click Confirm to accept the change. To configure intercom via phone user interface: Press Menu->Features->Intercom. Press , or the Switch soft key to select the desired values from the Accept Intercom, Intercom Mute, Intercom Tone and Intercom Barge fields.

  • Page 297: Configuring Advanced Features

    Configuring Advanced Features This chapter provides information for making configuration changes for the following advanced features: Distinctive Ring Tones  Tones  Remote Phone Book  LDAP  Busy Lamp Field  BLF List  Hide Features Access Code  Automatic Call Distribution ...

  • Page 298: Distinctive Ring Tones

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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.
  • Page 299 Configuring Advanced Features Maximu Minimum Nominal Bellcore Pattern Caden Pattern Duration Duration Tone Duratio (ms) (ms) n (ms) Silent Ringin Long 1025 Silent 2975 4000 4400 Ringin Short Silent Ringin Long 1000 1100 Bellcore-dr4 Silent Ringin Short Silent 2975 4000 4400 Ringin Bellcore-dr5...

  • Page 300: Auto Answer

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Value of N Ring Tone Ring7.wav Ring8.wav Silent.wav Splash.wav N<1 or N>10 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”...
  • Page 301 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 302 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default X ranges from 1 to 3 (for SIP-T23P/G). X ranges from 1 to 2 (for SIP-T21(P) E2). Web User Interface: Account->Advanced->Distinctive Ring Tones Phone User Interface: None features.alert_info_tone 0 or 1 Description: Enables or disables the IP phone to map the keywords in the Alert-info header to the specified Bellcore ring tones.
  • Page 303 Configuring Advanced Features Parameters Permitted Values Default 2-Ring2.wav 3-Ring3.wav 4-Ring4.wav 5-Ring5.wav 6-Ring6.wav 7-Ring7.wav 8-Ring8.wav 9-Silent.wav 10-Splash.wav Web User Interface: Settings->Ring->Internal Ringer File Phone User Interface: None To configure distinctive ring tones via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced.

  • Page 304: Tones

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones To configure the internal ringer text and internal ringer file via web user interface: Click on Settings->Ring. Enter the keywords in the Internal Ringer Text fields. Select the desired ring tones for each text from the pull-down lists of Internal Ringer File.
  • Page 305 Configuring Advanced Features Great Britain  Greece  Hungary  Lithuania  India  Italy  Japan  Mexico  New Zealand  Netherlands  Norway  Portugal  Spain  Switzerland  Sweden  Russia  United States  Chile ...
  • Page 306 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure Tones can be configured using the configuration files or locally. Configure the tones for the IP phone. Parameters: voice.tone.country voice.tone.dial voice.tone.ring voice.tone.busy Configuration File <y0000000000xx>.cfg 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.
  • Page 307 Configuring Advanced Features Parameters Permitted Values Default Settings->Tones->Select Country Phone User Interface: None 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.
  • Page 308 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default voice.tone.busy String Blank 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 parameter “voice.tone.country” is set to Custom. Web User Interface: Settings->Tones->Busy Phone User Interface:...
  • Page 309 Configuring Advanced Features Parameters Permitted Values Default 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 parameter “voice.tone.country” is set to Custom. Web User Interface: Settings->Tones->Dial Recall Phone User Interface:...
  • Page 310 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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”. Note: It works only if the parameter “voice.tone.country”...

  • Page 311: Remote Phone Book

    Configuring Advanced Features If you select Custom, you can customize a tone for each condition of the IP phone. Click Confirm to accept the change. Remote phone book is a centrally maintained phone book, stored on the remote server. Users only need the access URL of the remote phone book. The IP phone can establish a connection with the remote server and download the phone book, and then display the remote phone book entries on the phone user interface.
  • Page 312 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Specify how often the IP phone refreshes the local cache of the remote phone book. Parameter: features.remote_phonebook.flash_time Specify whether to refresh the local cache of the remote phone book at a time when accessing the remote phone book. features.remote_phonebook.enter_updat e_enable Specify the access URL of the remote...
  • Page 313 Configuring Advanced Features Parameters Permitted Values Default (X ranges from 1 to 5) Description: Configures the display name of the remote phone book item. Example: remote_phonebook.data.1.name = Xmyl Web User Interface: Directory->Remote Phone Book->Display Name Phone User Interface: None String within 99 Blank remote_phonebook.display_name characters...
  • Page 314 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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.

  • Page 315: Ldap

    Configuring Advanced Features To configure Incoming/Outgoing Call Lookup and Update Time Interval via web user interface: Click on Directory->Remote Phone Book. Select the desired value from the pull-down list of Incoming/Outgoing Call Lookup. Enter the desired time in the Update Time Interval (Seconds) field. Click Confirm to accept the change.
  • Page 316 Mobile or cellular phone number ipPhone IPphoneNumber Home phone number LDAP Phonebook on Yealink IP Phones For more information on LDAP , refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Procedure LDAP can be configured using the configuration files or locally.
  • Page 317 Configuring Advanced Features ldap.version ldap.call_in_lookup ldap.call_out_lookup ldap.ldap_sort Assign an LDAP key. Parameters: linekey.X.type/ programablekey.X.type Configure LDAP . Navigate to: http://<phoneIPAddress>/ servlet?p=contacts-LDAP &q=load Web User Interface Assign an LDAP key. Local Navigate to: http://<phoneIPAddress>/ servlet?p=dsskey&q=loa d&model=0 Phone User Interface Assign an LDAP key. Details of Configuration Parameters: Parameters Permitted Values...
  • Page 318 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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.
  • Page 319 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...
  • Page 320 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:...
  • Page 321 Configuring Advanced Features Parameters Permitted Values Default Directory->LDAP->Max Hits (1~32000) Phone User Interface: None String within 99 ldap.name_attr Blank characters 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.
  • Page 322 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Web User Interface: Directory->LDAP->LDAP Display Name Phone User Interface: None ldap.version 2 or 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:...
  • Page 323 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 324 Default Example: linekey.1.type = 38 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/T29G 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.
  • Page 325 Configuring Advanced Features Parameters Permitted Values Default 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).
  • Page 326 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default When X=9, the default value is 33 (Status). When X=10, the default value is 0 (NA). When X=14, the default value is 2 (Forward). Web User Interface: DSSKey->Line Key/ Programable Key->Type Phone User Interface: Menu->Features->DSS Keys->Line Key X->Type To configure LDAP via web user interface:...

  • Page 327: Busy Lamp Field

    Configuring Advanced Features (Optional.) Enter the string that will appear on the LCD screen in the Label field. Click Confirm to accept the change. To configure an LDAP key via phone user interface: Press Menu->Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field.
  • Page 328 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 configuration is only applicable to SIP-T46G/T42G/T41P/T29G/T27P/T23P/T23G/T21(P) E2 IP phones. BLF LED mode is also applicable to the expansion module EXP40 connected to SIP-T48G/T46G IP phones, EXP39 connected to SIP-T29G and SIP-T27P IP phones.
  • Page 329 Configuring Advanced Features 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 Description Fast flashing red (200ms) The monitored user receives an incoming call. The monitored user is dialing. The monitored user is talking.
  • Page 330 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure BLF can be configured using the configuration files or locally. 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: Configuration File y0000000000xx.cfg linekey.X.type linekey.X.line linekey.X.value...
  • Page 331 Configuring Advanced Features Parameters Permitted Values Default 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...
  • Page 332 Configures a DSS key as 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 333 1-16 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 334 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 335 Configuring Advanced Features (Optional.) Enter the string that will appear on the LCD screen in the Label field. (Optional.) Enter the directed call pickup code in the Extension field. Click Confirm to accept the change. To configure visual alert and audio alert for BLF pickup via web user interface: Click on Features->Call Pickup.

  • Page 336: Blf List

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired value from the pull-down list of BLF LED Mode. Click Confirm to accept the change. To configure a BLF key via phone user interface: Press Menu->Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select BLF from the Type field.
  • Page 337 Configuring Advanced Features Procedure BLF List can be configured using the configuration files or locally. 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: Configuration File y0000000000xx.cfg phone_setting.auto_blf_list_enable Configure the order of BLF list keys assigned automatically.
  • Page 338 Default 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 339 Description: Configures the call park retrieve code for account X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 340 0 or 1 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/T29G/T27P IP phones. Web User Interface: None Phone User Interface: None BLF List Key...
  • Page 341 1-16 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 342 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 1-Line 1 2-Line 2 … 16-Line 16 Example: linekey.1.line = 1 Web User Interface: DSSKey->Line Key->Line Phone User Interface: Menu->Features->DSS Keys->Line Key X->Account ID To configure the BLF List settings via web user interface: Click on Account.

  • Page 343: Hide Features Access Code

    Configuring Advanced Features Click Confirm to accept the change. To configure BLF List keys manually via web user interface: Click on DSSKey->Line Key (or Programable Key). In the desired DSS key field, select BLF List from the pull-down list of Type. Repeat step 2-3, configure more BLF list keys.
  • Page 344 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Configure the hide feature access codes feature. Local Web User Interface Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Details of Configuration Parameters: Permitted Parameters Default Values features.hide_feature_access_codes.enable 0 or 1 Description: Enables or disables the IP phone to display feature name instead of the feature access code when dialing and in talk.

  • Page 345: Automatic Call Distribution

    Configuring Advanced Features Select Enabled from the pull-down list of Hide Feature Access Codes. 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.
  • Page 346 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure ACD can be configured using the configuration files or locally. Configure ACD feature for account: Parameters: <MAC>.cfg account.X.acd.enable account.X.acd.available Assign an ACD key. Configuration File Parameters: linekey.X.type <y0000000000xx>.cfg Configure ACD auto available. Parameters: acd.auto_available acd.auto_available_timer Assign an ACD key.
  • Page 347 Enables or disables ACD feature for account X. 0-Disabled 1-Enabled X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 348 Parameters Permitted Values Default 1-Enabled X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G). X ranges from 1 to 2 (for SIP-T21(P) E2).
  • Page 349 Configuring Advanced Features Parameters Permitted Values Default For SIP-T46G/T29G 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.

  • Page 350: Message Waiting Indicator

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Enter the desired timer in the ACD Auto Available Timer (0~120s) field. Click Confirm to accept the change. To configure an ACD key via phone user interface: Press Menu->Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select ACD from the Type field.
  • Page 351 Enables or disables the IP phone to subscribe the message waiting indicator for account X. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will send a SUBSCRIBE message to the server for message-summary updates. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G).
  • Page 352 84600 Description: Configures MWI subscribe expiry time (in seconds) for account X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 353 Description: Configures the voice mail number for account X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 354 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G). X ranges from 1 to 2 (for SIP-T21(P) E2). Web User Interface: Account->Advanced->Voice Mail Display Phone User Interface:...

  • Page 355: Multicast Paging

    Configuring Advanced Features Enter the desired voice number in the Voice Mail field. Click Confirm to accept the change. To configure the presentation of audio and visual MWI via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced.
  • Page 356 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Users can send an RTP stream without involving SIP signaling by pressing a configured multicast paging key or a 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.
  • Page 357 Configuring Advanced Features 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 Phone User Interface...
  • Page 358 Configures a DSS key as a multicast paging key on the IP phone. The digit 24 stands for the key type Multicast Paging. X ranges from 1 to 29 (for SIP-T48G) X ranges from 1 to 27 (for SIP-T46G/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 359 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 360 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 361 For SIP-T23P/T23G/T21(P) E2 IP phones: The default value is 15. 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).
  • Page 362 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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).
  • Page 363 Configuring Advanced Features Parameters Permitted Values Default Menu->Features->DSS Keys->Line Key X->Type To configure a multicast paging key via web user interface: Click on DSSKey->Line Key. In the desired DSS key field, select Multicast Paging from the pull-down list of Type. Enter the multicast IP address and port number in the Value field.
  • Page 364 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired codec from the pull-down list of Multicast Codec. Click Confirm to accept the change. To configure two sending multicast addresses via web user interface: Click on Directory->Multicast IP. Enter the sending multicast address and port number in the Paging Address field. Enter the label in the Label field.
  • Page 365 Configuring Advanced Features Click Confirm to accept the change. To configure a multicast paging key via phone user interface: Press Menu->Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field. Press , or the Switch soft key to select Multicast Paging from the Key Type field.
  • Page 366 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure Configuration changes can be performed using the configuration files or locally. Configure the listening multicast address. Parameters: multicast.listen_address.X.ip_address <y0000000000xx>.cf multicast.listen_address.X.label Configuration File Configure Paging Barge and Paging Priority Active features. Parameters: multicast.receive_priority.enable multicast.receive_priority.priority Configure the listening multicast address.
  • Page 367 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the label to be displayed on the LCD screen when receiving the RTP multicast. Example: multicast.listen_address.1.label = Paging1 Web User Interface: Directory->Multicast IP->Label(Multicast Listening) Phone User Interface: None multicast.receive_priority.enable 0 or 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.
  • Page 368 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Enter the listening multicast address and port number in the Listening Address field. 1 is the highest priority and 10 is the lowest priority. Enter the label in the Label field. The label will appear on the LCD screen when receiving the RTP multicast. Click Confirm to accept the change.

  • Page 369: Call Recording

    Call-ID: 0_1289812066@10.3.20.14 CSeq: 2 INFO Contact: <sip:1009@10.3.20.14:5060> Max-Forwards: 70 User-Agent: Yealink SIP-T23G 44.80.0.20 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.

  • Page 370: Url Record

    GET /URLRecord/record.xml HTTP/1.1\r\n Request Method: GET Request URI: /URLRecord/record.xml Request version: HTTP/1.1 Host: 10.3.5.97:8080\r\n User-agent: Yealink SIP-T23G 44.80.0.20 00:15:65:74:B1:50\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>...
  • Page 371 Configures a DSS key as 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 372 Description: Configures a DSS key as 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/T29G)
  • Page 373 Menu->Features->DSS Keys->Line Key X->Type String within 99 linekey.X.value Blank characters 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P).
  • Page 374 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default X ranges from 1 to 21 (for SIP-T27P). X ranges from 1 to 3 (for SIP-T23P/G) X ranges from 1 to 2 (for SIP-T21(P) E2). Example: linekey.1.value = http://10.3.5.97:8080/URLRecord/record.xml Web User Interface: DSSKey->Line Key->Value Phone User Interface: Menu->Features->DSS Keys->Line Key X->Value...

  • Page 375: Hot Desking

    Configuring Advanced Features (Optional.) Enter the string that will appear on the LCD screen in the Label field. Click Confirm to accept the change. To configure a record key via phone user interface: Press Menu->Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field.
  • Page 376 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P)
  • Page 377 For SIP-T23P/T23G/T21(P) E2 IP phones: The default value is 15. 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).
  • Page 378 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Permitted Parameters Default Values 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).

  • Page 379: Action Url

    Configuring Advanced Features field. (Optional.) Enter the string that will appear on the LCD screen in the Label field. Press the Save soft key to accept the change. Action URL allows IP phones to interact with web server applications by sending an HTTP or HTTPS GET request.
  • Page 380 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Event Description Hold When the IP phone places a call on hold. UnHold When the IP phone resumes a hold call. Held When a call of the IP phone is held. UnHeld When the IP phone resumes a held call. Mute When the IP phone mutes a call.
  • Page 381 Configuring Advanced Features The following table lists pre-defined variable values. Variable Value Description $mac The MAC address of the IP phone The IP address of the IP phone $model The IP phone model $firmware The firmware version of the IP phone The SIP URI of the current account when the IP phone $active_url places a call, receives an incoming call or establishes...
  • Page 382 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones action_url.setup_completed action_url.registered action_url.unregistered action_url.register_failed action_url.off_hook action_url.on_hook action_url.incoming_call action_url.outgoing_call action_url.call_established action_url.dnd_on action_url.dnd_off action_url.always_fwd_on action_url.always_fwd_off action_url.busy_fwd_on action_url.busy_fwd_off action_url.no_answer_fwd_on action_url.no_answer_fwd_off action_url.transfer_call action_url.blind_transfer_call action_url.attended_transfer_call action_url.hold action_url.unhold action_url.held action_url.unheld action_url.mute action_url.unmute action_url.missed_call action_url.call_terminated action_url.busy_to_idle action_url.idle_to_busy action_url.ip_change action_url.forward_incoming_call action_url.reject_incoming_call action_url.answer_new_incoming_c action_url.transfer_finished action_url.transfer_failed...
  • Page 383 Configuring Advanced Features action_url.setup_autop_finish Configure action URL. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet?p =features-actionurl&q=load Details of Configuration Parameters: Parameters Permitted Values Default action_url.setup_completed URL within 511 characters Blank Description: Configures the action URL the IP phone sends after startup. The value format is: http(s)://IP address of server/help.xml? variable name=variable value.
  • Page 384 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Description: Configures the action URL the IP phone sends after an account is registered. Example: action_url.registered = http://192.168.0.20/help.xml?IP=$ip 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.
  • Page 385 Configuring Advanced Features Parameters Permitted Values Default Example: action_url.off_hook = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Off Hook Phone User Interface: None action_url.on_hook URL within 511 characters Blank Description: Configures the action URL the IP phone sends when on hook. Example: action_url.on_hook = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->On Hook...
  • Page 386 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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=$ip Web User Interface: Features->Action URL->Established...
  • Page 387 Configuring Advanced Features Parameters Permitted Values Default None action_url.always_fwd_on URL within 511 characters Blank Description: Configures the action URL the IP phone sends when always forward feature is enabled. Example: action_url.always_fwd_on = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Open Always Forward Phone User Interface: None action_url.always_fwd_off...
  • Page 388 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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=$ip Web User Interface: Features->Action URL->Close Busy Forward Phone User Interface: None action_url.no_answer_fwd_on...
  • Page 389 Configuring Advanced Features Parameters Permitted Values Default action_url.transfer_call URL within 511 characters Blank Description: Configures the action URL the IP phone sends when performing a transfer. Example: action_url.transfer_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Transfer Call Phone User Interface: None action_url.blind_transfer_call URL within 511 characters Blank...
  • Page 390 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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=$ip Web User Interface: Features->Action URL->Hold Phone User Interface: None action_url.unhold URL within 511 characters Blank Description: Configures the action URL the IP phone sends when a call being hold is resumed.
  • Page 391 Configuring Advanced Features Parameters Permitted Values Default action_url.unheld = http://192.168.0.20/help.xml?IP=$ip Web User Interface: None 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=$ip Web User Interface: Features->Action URL->Mute Phone User Interface:...
  • Page 392 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Phone User Interface: None action_url.call_terminated URL within 511 characters Blank Description: Configures the action URL the IP phone sends when terminating a call. Example: action_url.call_terminated = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Terminated Phone User Interface: None...
  • Page 393 Configuring Advanced Features Parameters Permitted Values Default 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=$ip Web User Interface: Features->Action URL->IP Changed Phone User Interface: None action_url.forward_incoming_call...
  • Page 394 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default None action_url.answer_new_incoming_call URL within 511 characters Blank Description: Configures the action URL the IP phone sends when answering a new incoming call. Example: action_url.answer_new_incoming_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Answer New-In Call Phone User Interface: None action_url.transfer_finished...

  • Page 395: Action Uri

    Configuring Advanced Features Parameters Permitted Values Default action_url.setup_autop_finish Blank URL within 511 characters 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: Click on Features->Action URL.
  • Page 396 Cancel actions or reject incoming calls or mute or un-mute calls. 0-9/*/POUND Press the keypad (0-9, * or #). Press the line keys (for SIP-T48G, X=29, for SIP-T46G/T29G, X=27, for SIP-T42G/T41P , L1-LX X=15, for SIP-T27P , X=21, for SIP-T23P/G, X=3, for SIP-T21(P) E2, X=2).
  • Page 397 Configuring Advanced Features Variable Value Phone Action Reset Reset a phone. Perform a semi-attended/attended transfer ATrans=xxx to xxx. BTrans=xxx Perform a blind transfer to xxx. CALLEND End a call. 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.
  • Page 398 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones features.action_uri_limit_ip Specify the trusted IP address(es) for sending the action URI to the IP phone. Local Web User Interface Navigate to: http://<phoneIPAddress>/servlet? 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.
  • Page 399 Configuring Advanced Features 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. Click Confirm to accept the change.

  • Page 400: Server Redundancy

    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 server fails, or the connection between the IP phone and the server fails.
  • Page 401 Working Server: Server 1 is configured with the domain name of the working server. For example: yealink.pbx.com. DNS mechanism is used such that the working server is resolved to multiple SIP servers for failover purpose. The working server is deployed in redundant pairs, designated as primary and secondary servers.

  • Page 402: Phone Registration

    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. Server Redundancy on Yealink IP For more information on server redundancy, refer to Phones , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 403 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 404 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 405 Configuring Advanced Features Parameters Permitted Values Default X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G). X ranges from 1 to 2 (for SIP-T21(P) E2).
  • Page 406 “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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 407 0-Disabled 1-Enabled X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 408 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Configure parameters of SIP server 1 and SIP server 2 in the corresponding fields. Click Confirm to accept the change. To configure server redundancy for failover purpose via web user interface: Click on Account->Register. Select the desired account from the pull-down list of Account.
  • Page 409 A query. If no port is found through the DNS query, 5060 will be used. The following details the procedures of DNS query for the IP phone to resolve the domain name (e.g., yealink.pbx.com) of working server into the IP address, port and transport protocol. NAPTR (Naming Authority Pointer) First, the IP phone sends NAPTR query to get the NAPTR pointer and transport protocol.
  • Page 410 SRV query next. TCP will be used, targeted to a host determined by an SRV query of “_sip._tcp.yealink.pbx.com”. If the flag of the NAPTR record returned is empty, the IP phone will perform NAPTR query again according to the previous NAPTR query result.

  • Page 411: Sip Server Domain Name Resolution

    Configuring Advanced Features 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: Sends the INVITE request to the primary server.
  • Page 412 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/T29G). X ranges from 1 to 12 (for SIP-T42G).

  • Page 413: Static Dns Cache

    Configuring Advanced Features Parameters Permitted Values Default 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.
  • Page 414 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure Static DNS cache can be configured only using the configuration files. 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 <y0000000000 xx>.cfg dns_cache_srv.X.name dns_cache_srv.X.port dns_cache_srv.X.priority dns_cache_srv.X.target Configuration File 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...
  • Page 415 Configuring Advanced Features Parameters Permitted Values Default 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 S, A, U or P Blank (X ranges from 1 to 12) Description: Configures the flag of NAPTR record X.
  • Page 416 Domain name Blank (X ranges from 1 to 12) 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 String within 32...
  • Page 417 Domain name Blank (X ranges from 1 to 12) 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 Integer from 0 to 65535 (X ranges from 1 to 12) Description: Configures the port to be used in SRV record X.
  • Page 418 Domain name Blank (X ranges from 1 to 12) 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 Domain name...
  • Page 419 None dns_cache_a.X.name Domain name Blank (X ranges from 1 to 12) 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 IP address Blank (X ranges from 1 to 12) Description: Configures the IP address that the domain name in A record X maps to.
  • Page 420 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).

  • Page 421: Lldp

    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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 422 Seconds until data unit expires. End of LLDPDU Marks end of LLDPDU. Name assigned to the IP phone. System Name The default value is “yealink”. Description of the IP phone. System Description The default value is “yealink”. The supported and enabled capabilities Optional TLVs of the IP phone.
  • Page 423 Serial number of the IP phone. Number Inventory – Manufacturer name of the IP phone. Manufacturer Name The default value is “yealink”. Inventory – Model Model name of the IP phone. Name Asset ID Assertion identifier of the IP phone.
  • Page 424 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Details of Configuration Parameters: Parameters Permitted Values Default network.lldp.enable 0 or 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.
  • Page 425 Configuring Advanced Features Enter the desired time interval in the Packet Interval (1~3600s) field. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone. To configure LLDP feature via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->LLDP->LLDP Status.
  • Page 426 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Navigate to: http://<phoneIPAddress>/servle t?p=network-adv&q=load Phone User Interface Configure CDP feature. Details of Configuration Parameters: Parameters Permitted Values Default network.cdp.enable 0 or 1 Description: Enables or disables CDP feature on the IP phone. 0-Disabled 1-Enabled Note: If it is set to 1, the IP phone will attempt to determine its VLAN ID through CDP .

  • Page 427: Vlan

    Configuring Advanced Features Enter the desired time interval in the Packet Interval (1~3600s) field. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone. To configure CDP feature via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->CDP->CDP Status.
  • Page 428 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. VLAN Feature on Yealink IP Phones For more information on VLAN, refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 429 Configuring Advanced Features Configure DHCP VLAN discovery feature. Navigate to: http://<phoneIPAddress>/servlet?p=n etwork-adv&q=load Configure VLAN for the Internet port and PC port. Phone User Interface Configure DHCP VLAN discovery feature. Details of Configuration Parameters: Parameters Permitted Values Default network.vlan.internet_port_enable 0 or 1 Description: Enables or disables VLAN for the Internet (WAN) port.
  • Page 430 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default network.vlan.internet_port_priority Integer from 0 to 7 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.
  • Page 431 Configuring Advanced Features Parameters Permitted Values Default ->Network->VLAN->PC Port->VID network.vlan.pc_port_priority Integer from 0 to 7 Description: Configures VLAN priority for the PC (LAN) 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.
  • Page 432 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Phone User Interface: Menu->Settings->Advanced Settings (default password: admin) ->Network->VLAN->DHCP VLAN->Option network.vlan.vlan_change.enable 0 or 1 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.
  • Page 433 Configuring Advanced Features Select the desired value (0-7) from the pull-down list of Priority. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after a reboot. Click OK to reboot the phone. To configure VLAN for PC port via web user interface: Click on Network->Advanced.
  • Page 434 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones To configure DHCP VLAN discovery via web user interface: Click on Network->Advanced. In the VLAN block, select the desired value from the pull-down list of DHCP VLAN Active. Enter the desired option in the Option (128-254) field. The default option is 132.
  • Page 435 VPN files are: certificates (ca.crt and client.crt), key (client.key) and the configuration file (vpn.cnf) of the VPN client. For more information on how to package a OpenVPN Feature on Yealink IP Phones TAR file, refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 436 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Configure VPN feature and upload a TAR package to the IP phone. Web User Interface Navigate to: Local http://<phoneIPAddress>/servlet?p =network-adv&q=load Phone User Interface Configure VPN feature. Details of Configuration Parameters: Parameters Permitted Values Default network.vpn_enable 0 or 1 Description:...
  • Page 437 Configuring Advanced Features Click Upload to upload the TAR file. The web user interface prompts the message “Import config…”. In the VPN block, select the desired value from the pull-down list of Active. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone.

  • Page 438: Voice Quality Monitoring

    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 mechanisms for voice quality monitoring are supported by Yealink IP phones: RTCP-XR ...

  • Page 439: Vq-Rtcpxr

    Configuring Advanced Features Parameters Permitted Values Default phone_setting.rtcp_xr_report.enable 0 or 1 Description: Enables or disables the IP phone to periodically (every 5 seconds) send RTCP-XR packets to another participating phone during a call for call quality monitoring and diagnosing. 0-Disabled 1-Enabled Note: It works only if the parameter “voice.rtcp_xr.enable”...
  • Page 440 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure RTCP-XR can be configured using the configuration files or locally. 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 phone_setting.vq_rtcpxr_interval_period Configure the generation of alert packets. Parameters: phone_setting.vq_rtcpxr_moslq_threshold_war ning...
  • Page 441 Configuring Advanced Features 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 phone_setting.vq_rtcpxr_display_moscq.enabl Configure the central report collector. Parameters: <MAC>.cfg account.X.vq_rtcpxr.collector_name account.X.vq_rtcpxr.collector_server_host account.X.vq_rtcpxr.collector_server_port Configure VQ-RTCPXR. Configure the phone to display RTP status showing the voice quality report of the last call on the web user interface.
  • Page 442 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Configure the central report collector. Navigate to: http://<phoneIPAddress>/servlet?p=account- adv&q=load&acc=0 Details of Configuration Parameters: Permitted Parameters Default Values phone_setting.vq_rtcpxr.session_report.enable 0 or 1 Description: Enables or disables the IP phone to send a session quality report to the central report collector at the end of each call.
  • Page 443 Configuring Advanced Features Permitted Parameters Default Values 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.
  • Page 444 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Permitted Parameters Default Values 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 less than or equal to 500, the phone will send a waring alert quality report to the central report collector;...
  • Page 445 Configuring Advanced Features Permitted Parameters Default Values 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 Description: Enables or disables the voice quality data of the last call or current call to be displayed on the LCD screen.
  • Page 446 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Permitted Parameters Default Values 0-Disabled 1-Enabled Note: It works only if 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 Description: Enables or disables the phone to display Local User on the LCD screen.
  • Page 447 Configuring Advanced Features Permitted Parameters Default Values Description: Enables or disables the phone to display Local Codec on the LCD screen. 0-Disabled 1-Enabled Note: It works only if 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...
  • Page 448 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Permitted Parameters Default Values phone_setting.vq_rtcpxr_display_jitter_buffer_max.enable 0 or 1 Description: Enables or disables the phone to display JitteBufferMax on the LCD screen. 0-Disabled 1-Enabled Note: It works only if 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:...
  • Page 449 Configuring Advanced Features Permitted Parameters Default Values Phone User Interface: None phone_setting.vq_rtcpxr_display_round_trip_delay.enable 0 or 1 Description: Enables or disables the phone to display RoundTripDelay on the LCD screen. 0-Disabled 1-Enabled Note: It works only if 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 Phone User Interface:...
  • Page 450 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 451 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 452 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones To configure interval report for VQ RTCP-XR via web user interface: Click on Settings->Voice Monitoring. Select the desired value from the pull-down list of VQ RTCP-XR Interval Report. Enter the desired value in the Period for Interval Report field. Click Confirm to accept the change.
  • Page 453 Configuring Advanced Features Enter the desired value in the Critical threshold for Delay field. Click Confirm to accept the change. To configure RTP status displayed on the web page via web user interface: Click on Settings->Voice Monitoring. Select the desired value from the pull-down list of Display Report options on Web.
  • Page 454 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Click Confirm to accept the change. The RTP status will appear on the web user interface at the path: Status. To configure RTP status displayed on the LCD screen via web user interface: Click on Settings->Voice Monitoring. Select the desired value from the pull-down list of Display Report options on phone.
  • Page 455 Configuring Advanced Features column and then click The selected list appears in the Enabled column. Repeat step 2 to add more items to the Enabled column. To remove an item from the Enabled column, select the desired item and then click To adjust the display order of enabled items, select the desired item and then click The LCD screen will display the item(s) in the adjusted order.

  • Page 456: Quality Of Service

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Enter the port of the central report collector in the VQ RTCP-XR Collector port field. Click Confirm to accept the change. 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.

  • Page 457: Voice Qos

    Configuring Advanced Features 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.
  • Page 458 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters: network.qos.rtptos network.qos.signaltos Configure the DSCPs for voice packets and SIP packets. Local Web User Interface Navigate to: http://<phoneIPAddress>/se rvlet?p=network-adv&q=lo Details of Configuration Parameters: Parameters Permitted Values Default network.qos.rtptos Integer from 0 to 63 Description: Configures the DSCP for voice packets.

  • Page 459: Network Address Translation

    Configuring Advanced Features To configure DSCPs for voice packets and SIP packets via web user interface: Click on Network->Advanced. Enter the desired value in the Voice QoS (0~63) field. Enter the desired value in the SIP QoS (0~63) field. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot.
  • Page 460 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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.
  • Page 461 Note: It works only if the value of the parameter “sip.nat_stun.enable” is set to 1 (Enabled). X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 462 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default Phone User Interface: Menu->Settings->Advanced Settings (default password: admin) ->Network->NAT->STUN Server Integer from 1024 to sip.nat_stun.port 3478 65000 Description: Configures the port of the STUN server. Example: sip.nat_stun.port = 3478 Web User Interface: Network->Advanced->NAT->STUN Port Phone User Interface: Menu->Settings->Advanced Settings (default password: admin)

  • Page 463: X Authentication

    Configuring Advanced Features Click Confirm to accept the change. To configure NAT traversal and STUN server via web user interface: Click on Network->Advanced. In the NAT block, select the desired value from the pull-down list of Active. Enter the IP address or the domain name of the STUN server in the STUN Server field.
  • Page 464 IP phone is allowed to access resources located on the protected side of the network. IP phones support protocols EAP-MD5, EAP-TLS, EAP-PEAP/MSCHAPv2, EAP-TTLS/EAP-MSCHAPv2, EAP-PEAP/GTC, EAP-TTLS/EAP-GTC and EAP-FAST for 802.1X authentication. Yealink 802.1X Authentication For more information on 802.1X authentication, refer to 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.
  • Page 465 Configuring Advanced Features Parameters Permitted Values Default Configures the 802.1x authentication method. 0-Disabled 1-EAP-MD5 2-EAP-TLS 3-EAP-PEAP/MSCHAPv2 4-EAP-TTLS/EAP-MSCHAPv2 5-EAP-PEAP/GTC 6-EAP-TTLS/EAP-GTC 7-EAP-FAST 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->Settings->Advanced Settings (default password: admin)
  • Page 466 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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->Settings->Advanced Settings (default password: admin)
  • Page 467 Configuring Advanced Features Parameters Permitted Values Default Phone User Interface: None To configure the 802.1X authentication via web user interface: Click on Network->Advanced. In the 802.1x block, select the desired protocol from the pull-down list of 802.1x Mode. a) If you select EAP-MD5: 1) Enter the user name for authentication in the Identity field.
  • Page 468 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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 select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system.
  • Page 469 Configuring Advanced Features 4) Click Upload to upload the certificate. d) If you select EAP-TTLS/EAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system.
  • Page 470 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system.
  • Page 471 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. g) If you select EAP-FAST: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field.
  • Page 472 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone. To configure the 802.1X authentication via phone user interface after: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->802.1x Settings.
  • Page 473 Configuring Advanced Features 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 ...
  • Page 474 AddObject object defined on the CPE. This method is used to remove a particular instance DeleteObject of an object. Yealink TR-069 Technote For more information on TR-069, refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Procedure TR-069 can be configured using the configuration files or locally.
  • Page 475 Configuring Advanced Features Parameters Permitted Values Default Description: Enables or disables TR-069 feature. 0-Disabled 1-Enabled Web User Interface: Settings->TR069->Enable TR069 Phone User Interface: None String within 128 managementserver.username Blank characters Description: Configures the user name for the IP phone to authenticate with the ACS (Auto Configuration Servers).
  • Page 476 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default characters 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 String within 128 managementserver.connection_request_username Blank characters Description: Configures the user name for the IP phone to authenticate the incoming connection...
  • Page 477 Configuring Advanced Features Parameters Permitted Values Default 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 Integer from 5 to managementserver.periodic_inform_interval 4294967295 Description:...
  • Page 478 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Enter the user name and password authenticated by the IP phone in the Connection Request Username and Connection Request Password fields. Click Confirm to accept the change. IPv6 is the next generation network layer protocol, designed as a replacement for the current IPv4 protocol.
  • Page 479 Note Stateful DHCPv6 address assignment method feature is only applicable to SIP-T48G/T46G/T29G IP phones. 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.
  • Page 480 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Details of Configuration Parameters: Parameters Permitted Values Default network.ip_address_mode 0, 1 or 2 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.
  • Page 481 Configuring Advanced Features Parameters Permitted Values Default 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: None network.ipv6_internet_port.ip IPv6 address Blank Description: Configures the IPv6 address when the IP address mode is configured as IPv6 or IPv4 &...
  • Page 482 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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.
  • Page 483 (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/T29G IP phones. Web User Interface: Network->Advanced->ICMPv6 Status->Active Phone User Interface: None To configure IPv6 address assignment method via web user interface: Click on Network->Basic.
  • Page 484 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones If you mark the Static IP Address radio box, configure the IPv6 address and other configuration parameters in the corresponding fields.
  • Page 485 Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after a reboot. Click OK to reboot the phone. To configure SLAAC feature via web user interface (only applicable to SIP-T48G/T46G/T29G): Click on Network->Advanced.
  • Page 486 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones In the ICMPv6 Status block, select the desired value from the pull-down list of Active. Click Confirm to accept the change. To configure IPv6 address assignment method via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->WAN Port.

  • Page 487: Configuring Audio Features

    Configuring Audio Features This chapter provides information for making configuration changes for the following audio features: Headset Prior  Dual Headset  Audio Codecs  Acoustic Clarity Technology  Headset prior allows users to use headset preferentially if a headset is physically connected to the IP phone.

  • Page 488: Dual Headset

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default 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: Click on Features->General Information.
  • Page 489 Configuring Audio Features Procedure Dual headset can be configured using the configuration files or locally. Configure dual headset. Configuration File <y0000000000xx>.cfg Parameter: features.headset_training Configure dual headset. Navigate to: Local Web User Interface http://<phoneIPAddress>/se rvlet?p=features-general&q =load Details of the Configuration Parameter: Parameter Permitted Values Default...

  • Page 490: Audio Codecs

    The following table lists the audio codecs supported by each phone model: Phone Model Supported Audio Codecs Default Audio Codecs G722, PCMA, PCMU, G729, SIP-T48G/T46G/T42 G722, PCMA, PCMU, G726-16, G726-24, G726-32, G/T41P/T29G G729 G726-40, iLBC, G723_53, G723_63 G722, PCMA, PCMU, G729,...
  • Page 491 Configuring Audio Features The following table summarizes the supported audio codecs on IP phones: Codec Algorithm Reference Bit Rate Sample Packetization Rate Time 20ms G722 G.722 RFC 3551 64 Kbps 16 Ksps PCMA G.711 RFC 3551 64 Kbps 8 Ksps 20ms a-law 20ms...
  • Page 492 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Codec Configuration Methods Priority RTPmap Web User Interface Configuration Files G723_53 Web User Interface Configuration Files G723_63 Web User Interface Configuration Files G726-16 Web User Interface Configuration Files G726-24 Web User Interface Configuration Files G726-32 Web User Interface Configuration Files...
  • Page 493 Enables or disables the specified codec for account X. 0-Disabled 1-Enabled X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 494 1 to 11) Description: Configures the codec for account X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 495 1 to 9) Description: Configures the priority of the enabled codec for account X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 496 1 to 9) Description: Configures the rtpmap of the audio codec for account X. X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G).
  • Page 497 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/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P).
  • Page 498 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default account.1.ptime = 20 Web User Interface: Account->Advanced->PTime (ms) Phone User Interface: None To configure the codecs to use and adjust the priority of the enabled codecs on a per-line basis via web user interface: Click on Account.

  • Page 499: Acoustic Echo Cancellation

    Configuring Audio Features Select the desired value from the pull-down list of PTime (ms). 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.
  • Page 500 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones voice.echo_cancellation Configure AEC. Navigate to: Local Web User Interface http://<phoneIPAddress>/ servlet?p=settings-voice& q=load Details of the Configuration Parameter: Parameter Permitted Values Default voice.echo_cancellation 0 or 1 Description: Enables or disables AEC (Acoustic Echo Canceller) feature on the IP phone. 0-Disabled 1-Enabled Web User Interface:...

  • Page 501: Background Noise Suppression

    Configuring Audio Features 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.

  • Page 502: Comfort Noise Generation

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameter Permitted Values Default 0-Disabled 1-Enabled Web User Interface: Settings->Voice->Echo Cancellation->VAD Phone User Interface: None To configure VAD via web user interface: Click on Settings->Voice. Select the desired value from the pull-down list of VAD. Click Confirm to accept the change.
  • Page 503 Configuring Audio Features Procedure CNG can be configured using the configuration files or locally. Configure CNG. Configuration File <y0000000000xx>.cfg Parameter: voice.cng Configure CNG. Navigate to: Local Web User Interface http://<phoneIPAddress>/ servlet?p=settings-voice& q=load Details of the Configuration Parameter: Parameter Permitted Values Default voice.cng 0 or 1...

  • Page 504: Jitter Buffer

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired value from the pull-down list of CNG. 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.
  • Page 505 Configuring Audio Features Navigate to: http://<phoneIPAddress>/ servlet?p=settings-voice& q=load Details of Configuration Parameters: Parameters Permitted Values Default voice.jib.adaptive 0 or 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 Description: Configures the minimum delay time (in milliseconds) of jitter buffer.
  • Page 506 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default voice.jib.normal Integer from 0 to 400 Description: Configures the normal delay time (in milliseconds) of jitter buffer. Note: It works only if the parameter “voice.jib.adaptive” is set to 0 (Fixed). Web User Interface: Settings->Voice->JITTER BUFFER->Normal Phone User Interface:...

  • Page 507: Configuring Security Features

    Configuring Security Features This chapter provides information for making configuration changes for the following security-related features: Transport Layer Security  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.
  • Page 508 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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 ...
  • Page 509 A unique server certificate: It is unique to an IP phone (based on the MAC address) and issued by the Yealink Certificate Authority (CA). A generic server certificate: It issued by the Yealink Certificate Authority (CA). Only if no unique certificate exists, the IP phone may send a generic certificate for authentication.
  • Page 510 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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 In TLS feature, we use the terms trusted and server certificate. These are also known as CA and device certificates.
  • Page 511 0-UDP 1-TCP 2-TLS 3-DNS-NAPTR X ranges from 1 to 16 (for SIP-T48G/T46G/T29G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P/T27P). X ranges from 1 to 3 (for SIP-T23P/G). X ranges from 1 to 2 (for SIP-T21(P) E2).
  • Page 512 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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.
  • Page 513 Configuring Security Features Parameters Permitted Values Default Security->Trusted Certificates->Common Name Validation Phone User Interface: None security.dev_cert 0 or 1 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.
  • Page 514 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 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 phone_setting.reserve_certs_enable 0 or 1 Description: Enables or disables the IP phone to reserve custom certificates after it is reset to...
  • Page 515 Configuring Security Features To configure the trusted certificates via web user interface: Click on Security->Trusted Certificates. Select the desired values from the pull-down lists of Only Accept Trusted Certificates, Common Name Validation and CA Certificates. Click Confirm to accept the change. To upload a trusted certificate via web user interface: Click on Security->Trusted Certificates.

  • Page 516: Secure Real-Time Transport Protocol

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones To configure the server certificates via web user interface: Click on Security->Server Certificates. Select the desired value from the pull-down list of Device Certificates. Click Confirm to accept the change. To upload a server certificate via web user interface: Click on Security->Server Certificates.
  • Page 517 Configuring Security Features 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 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.
  • Page 518 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/T29G). X ranges from 1 to 12 (for SIP-T42G).

  • Page 519: Encrypting Configuration Files

    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...

  • Page 520: Procedure To Encrypt Configuration Files

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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 Yealink Configuration Encryption Tool User Guide required.
  • Page 521 Configuring Security Features (Optional.) Click Browse to locate the target directory from your local system in the Target Directory field. The tool uses the file folder “Encrypted” as the target directory by default. (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.
  • Page 522 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure Decryption method can be configured using the configuration files. Configure the decryption method. Parameter: auto_provision.aes_key_in_file Configure AES keys. Configuration File <y0000000000xx>.cfg Parameters: auto_provision.aes_key_16.com auto_provision.aes_key_16.mac auto_provision.update_file_mode Configure AES keys. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet?p =settings-autop&q=load Details of Configuration Parameters:...
  • Page 523 Configuring Security Features 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: # $ % * + , - .
  • Page 524 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default 1-Enabled Web User Interface: None Phone User Interface: None To configure AES keys via web user interface: Click on Settings->Auto Provision. 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: # $ % * + , - .

  • Page 525: Resource Files

    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):...

  • Page 526: Replace Rule Template

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Directory Template  Super Search Template  Local Contact File  Remote XML Phone Book  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.

  • Page 527: Dial-Now Template

    Resource Files <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.

  • Page 528: Softkey Layout Template

    The softkey layout template allows you to customize soft key layout for 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.

  • Page 529: Directory Template

    Resource Files <Disable> <Key Type="Empty"/> <Key Type="End Call"/> </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> Directory provides easy access to frequently used lists. Users can access lists by pressing the Dir soft key when the IP phone is idle.

  • Page 530: Super Search Template

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Procedure Use the following procedures to customize a directory template. Customizing a directory template: Open the template file using an ASCII editor. For each directory list that you want to configure, edit the corresponding string in the file.
  • Page 531 Resource Files <root_super_search> indicates the start of a template and </root_super_search>  indicates the end of a template. The default display names of the 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 ...
  • Page 532 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 only applicable to SIP-T48G, SIP-T46G and SIP-T29G IP phones. At most 48 groups can be added to the IP phone. ...

  • Page 533: Local Contact File

    Resource Files Procedure Use the following procedures to customize a local contact template file. To customize a local contact file: Open the template file using an ASCII editor. For each group that you want to add, add the following string to the file. Each starts on a separate line: <group display_name=""...

  • Page 534: Remote Xml Phone Book

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones </root_contact> 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).
  • Page 535 Specify the key between <Name> and </Name>. Specify the access URL of a XML file between </URL> and </URL>. 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>...
  • Page 536 <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 phonebook generation tool to generate a remote XML phone book. Yealink Phonebook Generation Tool User Guide For more information, refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.

  • Page 537: Troubleshooting

    Troubleshooting This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using IP phones. IP phones can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which can help an administrator more easily find the system problem and fix it.
  • Page 538 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones overwritten or appended. syslog.ftp.max_logfile - Specify the maximum size of the log files on the  provisioning server. syslog.ftp.append_limit_mode - Specify the phone to stop log upload or delete the  old lod when the log on the provisioning server reaches the max size. syslog.bootlog_upload_wait_time - Specify the waiting time before the phone ...
  • Page 539 Troubleshooting log files on the provisioning server. Parameters: syslog.ftp.max_logfile Configures the phone to stop log upload or delete the old log when the log on the provisioning server reaches the max size. Parameters: syslog.ftp.append_limit_mode Configures the waiting time before the phone uploads the log file to the provisioning server.
  • Page 540 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones http://<phoneIPAddress>/servlet?p =settings-config&q=load Details of Configuration Parameters: Parameters Permitted Values Default syslog.mode 0, 1 or 2 Description: Configures the IP phone to export log files to an FTP/TFTP server (provisioning server), syslog server or the local system. 0-Local 1-Syslog Server 2-FTP/TFTP Server...
  • Page 541 Troubleshooting Parameters Permitted Values Default Example: syslog.log_upload_period = 60 Note: It works only if the parameter “syslog.mode” is set to 2 (FTP/TFTP Server). If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Settings->Configuration->Upload Period(30~2592000)s Phone User Interface: None...
  • Page 542 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Parameters Permitted Values Default None syslog.ftp.append_limit_mode 1 or 2 Description: Configures the phone to stop log upload or delete the old log when the log on the provisioning server reaches the max size. 1-Append Delete 2-Append Stop Note: It works only if the parameter “syslog.mode”...
  • Page 543 Troubleshooting Parameters Permitted Values Default 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 Web User Interface: Settings->Configuration->System Log Level Phone User Interface: None To configure the level of the system log via web user interface: Click on Settings->Configuration.
  • Page 544 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones To configure the phone to export the system log to a syslog server via web user interface: Click on Settings->Configuration. Mark the Server radio box in the Export System Log field. Enter the IP address or domain name of the syslog server in the Server Name field. Click Confirm to accept the change.
  • Page 545 Troubleshooting Click Export to open file download window, and then save the file to your local system. The following figure shows a portion of a log file- an account registration: To configure the phone to export the system log to an FTP/TFTP server via web user interface: Click on Settings->Configuration.
  • Page 546 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Select the desired limit mode from the pull-down list of Append Limit Mode. 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.
  • Page 547 Troubleshooting The following figure shows a portion of a <mac>-sys.log (e.g., 0015654146dd-sys.log): You can capture packet in two ways: capturing the packet via web user interface or using the Ethernet software. You can analyze the packet captured for troubleshooting purpose. To capture packets via web user interface: Click on Settings->Configuration.
  • Page 548 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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.
  • Page 549 Troubleshooting Parameter Permitted Values Default Settings->Preference->WatchDog Phone User Interface: None To configure watch dog feature via web user interface: Click on Settings->Preference. Select the desired value from the pull-down list of WatchDog. Click Confirm to accept the change. Status indicators may consist of the power LED, MESSAGE key LED, line key indicator, headset key indicator and the on-screen icon.
  • Page 550 The <mac>-local.cfg configuration file contains changes made via phone user interface and web user interface. The config.bin file is an encrypted file. For more information on config.bin file, contact your Yealink reseller. To export BIN configuration files via web user interface: Click on Settings->Configuration.
  • Page 551 Troubleshooting Click Export to open file download window, and then save the file to your local system. To import a BIN configuration file via web user interface: Click on Settings->Configuration. In the Export or Import Configuration block, click Browse to locate a BIN configuration file from your local system.
  • Page 552 Click Import to import the configuration file. 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.
  • Page 553 Troubleshooting 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. ...
  • Page 554 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones If you have poor sound quality/acoustics like intermittent voice, low volume, echo or other noises, the possible reasons could be: Users are seated too far out of recommended microphone range and sound faint,  or are seated too close to sensitive microphones and cause echo.
  • Page 555 (the size of the uncompressed file compared to that of the compressed file) and can be stored in smaller space. Tools for converting BMP format to DOB format are available. For more information, refer to Yealink IP Phones Auto Provisioning Guide available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 556 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones able to download the CFG 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. ...
  • Page 557 Troubleshooting To reset the IP phone via web user interface: Click on Settings->Upgrade. Click Reset to Factory Setting in the Reset to Factory Setting field. The web user interface prompts the message “Do you want to reset to factory?”. Click OK to confirm the resetting. The IP phone will be reset to factory sucessfully after startup.
  • Page 558 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones...

  • Page 559: Appendix

    Appendix 802.1x--an IEEE Standard for port-based Network Access Control (PNAC). It is a part of the IEEE 802.1 group of networking protocols. It provides an authentication mechanism to devices wishing to attach to a LAN or WLAN. ACS (Auto Configuration server)--responsible for auto-configuration of the Central Processing Element (CPE).
  • Page 560 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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.
  • Page 561 Appendix Time Zone Time Zone Name Samoa US-Hawaii-Aleutian, US-Alaska-Aleutian -9:30 French Polynesia US-Alaska Time Canada(Vancouver,Whitehorse), Mexico(Tijuana,Mexicali), US-Pacific Time Canada(Edmonton,Calgary), Mexico(Mazatlan,Chihuahua), US-MST no DST, US-Mountain Time Canada-Manitoba(Winnipeg), Chile(Easter Islands), Mexico(Mexico City,Acapulco), US-Central Time Bahamas(Nassau), Canada(Montreal,Ottawa,Quebec), Cuba(Havana), US-Eastern Time -4:30 Venezuela(Caracas) Canada(Halifax,Saint John), Chile(Santiago), Paraguay(Asuncion), UK(Falkland Islands), UK-Bermuda(Bermuda), Trinidad&Tobago -3:30...
  • Page 562 New Zealand(Wellington,Auckland), Russia(Kamchatka Time) +12:45 New Zealand(Chatham Islands) Tonga(Nukualofa) +13:30 Chatham Islands Kiribati Yealink IP phones trust the following CAs by default: DigiCert High Assurance EV Root CA  Deutsche Telekom AG Root CA-2  Equifax Secure Certificate Authority ...
  • Page 563  Note Yealink endeavors to maintain a built-in list of most common used CA Certificates. Due to memory constraints, we cannot ensure a complete set of certificates. If you are using a certificate from a commercial Certificate Authority not in the list above, you can send a request to your local distributor.
  • Page 564 Configures 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/T29G) X ranges from 1 to 15 (for SIP-T42G/T41P) X ranges from 1 to 21 (for SIP-T27P)
  • Page 565 Appendix  Direct Pickup  Call Park  DTMF  Voice Mail  Speed Dial  Intercom  Line    Group Listening  XML Group  Group Pickup  Multicast Paging  Record  XML Browser  Hot Desking ...
  • Page 566  Format Integer 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/T29G 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.
  • Page 567 Appendix The default value is 15. 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).
  • Page 568 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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).
  • Page 569 Appendix 11-DTMF 12-Voice Mail 13-Speed Dial 14-Intercom 15-Line 16-BLF 17-URL 18-Group Listening 22-XML Group 23-Group Pickup 24-Multicast Paging 25-Record 27-XML Browser 28-History 30-Menu 32-New SMS 33-Status 34-Hot Desking 35-URL Record 38-LDAP 39-BLF List 40-Prefix 41-Zero Touch 42-ACD 43-Local Directory 45-Local Group 47-XML Directory 50-Phone Lock 51-Switch Account Up...
  • Page 570 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/T29G). X ranges from 1 to 15 (for SIP-T42G/T41P). X ranges from 1 to 21 (for SIP-T27P).
  • Page 571 16-Line 16 Example linekey.1.line = 2 Configuration File Parameter- <y0000000000xx>.cfg linekey.X.value Parameter- programablekey.X.value Configures the value for some key features. For line keys: Description X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G/T29G).
  • Page 572 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/T29G). Description X ranges from 1 to 15 (for SIP-T42G/T41P).
  • Page 573 <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/T29G). Description X ranges from 1 to 15 (for SIP-T42G/T41P).
  • Page 574 3-GroupCommon 4-EnterpriseCommon 5-Personal Format Integer Default Value Range 0 to 48 Configures the second remote phone book. Example linekey.1.xml_phonebook = 1 This section describes how Yealink IP phones comply with the IETF definition of SIP as described in RFC 3261.

  • Page 575: Rfc And Internet Draft Support

    Appendix This section contains compliance information in the following: RFC and Internet Draft Support  SIP Request  SIP Header  SIP Responses  SIP Session Description Protocol (SDP) Usage  The following RFC’s and Internet drafts are supported: RFC 1321—The MD5 Message-Digest Algorithm ...
  • Page 576 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 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) ...
  • Page 577 Appendix 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 ...

  • Page 578: Sip Request

    RFC number. The following SIP request messages are supported: Method Supported Notes REGISTER Yealink IP phones support mid-call changes such as placing a call on hold as INVITE signaled by a new INVITE that contains an existing Call-ID.
  • Page 579 Appendix Method Supported Notes Accept Alert-Info Allow Allow-Events Authorization Call-ID Call-Info Contact Content-Length Content-Type CSeq Diversion History-Info 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 Require Route...

  • Page 580: Sip Responses

    Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Method Supported Notes RSeq Session-Expires Subscription-State Supported User-Agent The following SIP responses are supported: In the following table, a “Yes” in the Supported column means the header is sent and Note properly parsed. The phone may not actually generate the response. 1xx Response—Information Responses 1xx Response Supported...
  • Page 581 Appendix 3xx Response Supported Notes 301 Moved Permanently 302 Moved Temporarily 4xx Response—Request Failure Responses 4xx Response Supported Notes 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...
  • Page 582 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones 4xx Response Supported Notes 485 Ambiguous 486 Busy Here 487 Request Terminated 488 Not Acceptable Here 491 Request Pending 493 Undecipherable 5xx Response—Server Failure Responses 5xx Response Supported Notes 500 Internal Server Error 501 Not Implemented 502 Bad Gateway 503 Service Unavailable 504 Gateway Timeout...
  • Page 583 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.
  • Page 584 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones The call flow scenario is as follows: User A calls User B. User B answers the call. User B hangs up. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 100 Trying F4.
  • Page 585 Appendix Step 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 ...
  • Page 586 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: User A calls User B.
  • Page 587 Appendix The call cannot be set up successfully. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 100 Trying F4. 100 Trying F5. 486 Busy Here F6. 486 Busy Here F7. ACK F8. ACK Step Action Description User A sends the INVITE message to a proxy server.
  • Page 588 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.
  • Page 589 Appendix The call flow scenario is as follows: User A calls User B. User B does not answer the call. User A hangs up. The call cannot be set up successfully. User A Proxy Server User B F1. INVITE B F2.
  • Page 590 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Step Action Description 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 INVITE—Proxy Server to User To field to User B.

  • Page 591: Successful Call Setup And Call Hold

    Appendix 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: User A calls User B.
  • Page 592 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Step 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 ...
  • Page 593 Appendix Step Action Description connection has been made. The proxy server forwards the 200 OK 200 OK—Proxy Server to User 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.
  • Page 594 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: User A calls User B.
  • Page 595 Appendix User B accepts the call from User C. Proxy Server User C User A 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.
  • Page 596 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Step 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 ...
  • Page 597 Appendix Step Action Description call session is now active. The proxy server sends the SIP ACK to User B. The ACK confirms that the proxy ACK—Proxy Server to User B server has received the 200 OK response. The call session is now active. User C sends a SIP INVITE message to the proxy server.
  • Page 598 The proxy server forwards the SIP ACK to User A to confirm that User C has ACK—Proxy Server to User A received the 200 OK response. The following figure illustrates a successful call between Yealink SIP IP phones in which...
  • Page 599 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.
  • Page 600 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Call is established between User A and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7.
  • Page 601 Appendix Step Action Description User A sends an INVITE message to the proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
  • Page 602 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to User B.
  • Page 603 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.
  • Page 604 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones User A transfers the call to User C. Call is established between User B and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4.
  • Page 605 Appendix Step 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 ...
  • Page 606 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to User B.
  • Page 607 Appendix Step Action Description sends the INVITE request to User C. User C sends a SIP 180 Ringing 180 Ringing—User C to Proxy response to the proxy server. The 180 Server Ringing response indicates that the user is being alerted. The proxy server forwards the 180 180 Ringing—Proxy Server to Ringing response to User A.

  • Page 608: Always Call Forward

    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.
  • Page 609 Appendix User C answers the call. Call is established between User A and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 302 Move Temporarily F4. ACK F5. 302 Move Temporarily F6. ACK F7.
  • Page 610 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of the User B is ...

  • Page 611: Busy Call Forward

    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...
  • Page 612 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: User B enables busy call forward, and the destination number is User C.
  • Page 613 Appendix Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
  • Page 614 ACK— User A to Proxy Server has received the 200 OK response. The call session is now active. The proxy server sends the ACK ACK—Proxy Server to User C message to User C. The following figure illustrates successful call forwarding between Yealink SIP IP phones...

  • Page 615: No Answer Call Forward

    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.
  • Page 616 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Step 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 ...
  • Page 617 The proxy server sends the ACK message to User C. The ACK confirms ACK—Proxy Server to User C that the proxy server has received the 200 OK response. The following figure illustrates successful 3-way calling between Yealink IP phones in...
  • Page 618 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: User A calls User B.
  • Page 619 Appendix Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
  • Page 620 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Step Action Description connection has been made. User A sends a SIP ACK to the proxy server. The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active.
  • Page 621 Appendix Step Action Description The proxy server maps the SIP URI in the INVITE—Proxy Server to User To field to User C. The proxy server sends the SIP INVITE request to User C. User C sends a SIP 180 Ringing 180 Ringing—User C to Proxy response to the proxy server.
  • Page 622 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones...
  • Page 623 Index Numeric 180 Ring Workaround Call Completion 802.1X Authentication Call Forward Call Hold Call Log Call Park About This Guide Call Recording Acoustic Echo Cancellation Call Transfer Action URL Call Waiting Action URI Calling Line Identification Presentation Administrator Password Connected Line Identification Presentation Always Forward Capturing Packets Analyzing the Configuration Files...
  • Page 624 Administrator’s Guide for SIP-T2_Series_T4_Series IP Phones Enabling the Watch Dog Feature Message Waiting Indicator Missed Call Log Feature Key Synchronization Multicast Paging Getting Information from Status Indicators NAT Traversal Getting Started Network Address Translation (NAT) Group Call Pickup Network Conference No Answer Forward Notification Popups H.323...
  • Page 625 Index SIP Components Web Server Type SIP Header Web User Interface SIP IP Phone Models SIP Request SIP Responses SIP Session Description Protocol Usage SIP Session Timer Softkey Layout Specifying the Language to Use SRTP STUN Server Suppress DTMF Display Table of Contents Time and Date Transfer on Conference Hang Up...

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