Yealink SIP-T4X Administrator's Manual
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Yealink SIP-T4X Administrator's Manual

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Summary of Contents for Yealink SIP-T4X

  • Page 2 Copyright © 2014 YEALINK NETWORK TECHNOLOGY Copyright © 2014 Yealink Network Technology CO., LTD. All rights reserved. No parts of this publication may be reproduced or transmitted in any form or by any means, electronic or mechanical, photocopying, recording, or otherwise, for any purpose, without the express written permission of Yealink Network Technology CO., LTD.
  • 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 SIP-T4X IP phone firmware contains third-party software under the GNU General Public License (GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms and conditions of the license. The original GPL license, source code of components licensed under GPL and used in Yealink products can be downloaded online: http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293.

  • Page 5: About This Guide

    BroadSoft features on the BroadWorks web portal and IP phones. For support or service, please contact your Yealink reseller or go to Yealink Technical Support online: http://www.yealink.com/Support.aspx. The information detailed in this guide is applicable to the firmware version 73 or higher.
  • Page 6 Administrator’s Guide for SIP-T4X IP Phones on IP phones. Chapter 4, “Configuring Advanced Features” describes how to configure  advanced features on IP phones. Chapter 5, “Configuring Audio Features” describes how to configure audio features  on IP phones. Chapter 6, “Configuring Security...
  • Page 7 About This Guide Call Display on page  Input Method Customization on page  Off Hook Hot Line Dialing on page  Feature Key Synchronization on page  BLF List on page  Capturing the Current Screen of the Phone on page ...
  • Page 8 Administrator’s Guide for SIP-T4X IP Phones Automatic Gain Control on page  Major updates have occurred to the following section: Configuration Methods on page  Call Hold on page  Audio Codecs on page  Acoustic Clarity Technology on page ...
  • Page 9 About This Guide DHCP on page  Backlight on page  Time and Date on page  Key as Send on page  Anonymous Call on page  Busy Lamp Field on page  Action URL on page  IPv6 Support on page ...
  • Page 10 Administrator’s Guide for SIP-T4X IP Phones Auto Answer on page  Audio Codecs on page  Encrypting Configuration Files on page  This version is updated to incorporate T41P as one of the T4X device models. The following section is new for this version:...

  • Page 11: Table Of Contents

    Product Overview ..............1 VoIP Principle ............................ 1 SIP Components..........................2 SIP IP Phone Models ........................3 Physical Features of SIP-T4X IP Phones ................... 4 Key Features of SIP-T4X IP Phones ................... 7 Getting Started ................. 9 Connecting the IP Phones ....................... 9 Initialization Process Overview ....................
  • Page 12 Administrator’s Guide for SIP-T4X IP Phones Setting up the Provisioning Server ..................19 Deploying Phones from the Provisioning Server ..............19 Configuring Basic Network Parameters ..................21 DHCP ............................21 Configuring Network Parameters Manually ................ 26 PPPoE ............................32 Configuring Transmission Methods of the Internet Port and PC Port ......... 35 Configuring PC Port ........................

  • Page 13: Table Of Contents

    Table of Contents Auto Answer ..........................149 Call Completion ........................... 152 Anonymous Call ........................... 154 Anonymous Call Rejection ......................158 Do Not Disturb ..........................162 Busy Tone Delay ........................... 172 Return Code When Refuse ......................174 Early Media ..........................175 180 Ring Workaround ........................
  • Page 14 Administrator’s Guide for SIP-T4X IP Phones Receiving RTP Stream ......................324 Call Recording ..........................328 Hot Desking ..........................334 Action URL ............................ 337 Action URI ............................. 354 Capturing the Current Screen of the Phone ............... 358 Server Redundancy ........................359 SIP Server Domain Name Resolution ..................
  • Page 15 Table of Contents Directory Template ........................485 Super Search Template ....................... 486 Local Contact File ........................488 Remote XML Phone Book ......................490 Troubleshooting ..............493 Troubleshooting Methods ......................493 Viewing Log Files ........................493 Capturing Packets ........................ 498 Enabling the Watch Dog Feature ..................499 Getting Information from Status Indicators ................
  • Page 16 Administrator’s Guide for SIP-T4X IP Phones SIP Responses ........................528 SIP Session Description Protocol (SDP) Usage ..............530 Appendix E: SIP Call Flows ......................531 Successful Call Setup and Disconnect ................531 Unsuccessful Call Setup—Called User is Busy ..............534 Unsuccessful Call Setup—Called User Does Not Answer ..........

  • Page 17: Product Overview

    Product Overview This chapter contains the following information about SIP-T4X 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 18 Administrator’s Guide for SIP-T4X IP Phones SIP provides capabilities to: Determine the location of the target endpoint -- SIP supports address resolution,  name mapping, and call redirection. Determine the media capabilities of the target endpoint -- Via Session Description ...
  • Page 19 SIP-T41P  SIP-T4X IP phones comply with the SIP standard (RFC 3261), and they can only be used within a network that supports this type of phone. In order to operate as SIP endpoints in your network successfully, SIP-T4X IP phones must meet the following requirements: A working IP network is established.

  • Page 20: Physical Features Of Sip-T4X Ip Phones

    Administrator’s Guide for SIP-T4X IP Phones A call server is active and configured to receive and send SIP messages.  This section lists the available physical features of SIP-T4X IP phones. SIP-T48G Physical Features: 7” 800 x 480 pixel color touch screen with backlight...
  • Page 21 Product Overview SIP-T46G Physical Features: 4.3” 480 x 272 pixel color display with backlight 24 bit depth color 16 VoIP accounts, Broadsoft Validated/Asterisk Compatible ® HD Voice: HD Codec, HD Handset, HD Speaker 40 keys including 10 line keys 1*RJ9 (4P4C) handset port 1*RJ9 (4P4C) headset port 2*RJ45 10/100/1000Mbps Ethernet ports 1*RJ12 (6P6C) expansion module port...
  • Page 22 Administrator’s Guide for SIP-T4X IP Phones SIP-T42G Physical Features: 192 x 64 graphic LCD 12 VoIP accounts, Broadsoft Validated/Asterisk ® Compatible HD Voice: HD Codec, HD Handset, HD Speaker 34 keys including 6 line keys 1*RJ9 (4P4C) handset port 1*RJ9 (4P4C) headset port...
  • Page 23 10 LEDs: 1*power, 6*line, 1*mute, 1*headset, 1*speakerphone Power adapter: AC 100~240V input and DC 5V/1.2A output Power over Ethernet (IEEE 802.3af) In addition to physical features introduced above, SIP-T4X IP phones also support the following key features when running the latest firmware: Phone Features ...
  • Page 24 Administrator’s Guide for SIP-T4X IP Phones identity, auto answer. Advanced Features: BLF , server redundancy, distinctive ring tones, remote phone book, LDAP , 802.1X authentication. Codecs and Voice Features  Wideband codec: G.722 Narrowband codec: G.711 (A/μ), G.723, G.726, G.729, iLBC.

  • Page 25: Getting Started

    Configuring Basic Network Parameters  Upgrading Firmware  This section introduces how to install SIP-T4X IP phones with the components in packaging contents. Attach the stand and optional wall mount bracket Connect the handset and optional headset Connect the network and power...
  • Page 26 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. Yealink Wall Mount For more information on how to mount the IP phone to a wall, refer to Quick Installation Guide for SIP-T4X IP Phones...
  • Page 27 For more information on how to use the EHS36 on the IP phone, refer to User Guide. Bluetooth can only be used on the SIP-T48G and SIP-T46G IP phones. For more information Yealink Bluetooth USB on how to use the Bluetooth on SIP-T48G/T46G IP phones, refer to Dongle BT40 User Guide...
  • Page 28 Administrator’s Guide for SIP-T4X IP Phones Connect the included or a standard Ethernet cable between the Internet port on IP phones and the one on the wall or switch/hub device port. Power over Ethernet With the included or a regular Ethernet cable, IP phones can be powered from a PoE-compliant switch or hub.
  • Page 29 26. Contacting the auto provisioning server SIP-T4X IP phones support the FTP , TFTP , HTTP , and HTTPS protocols for auto provisioning and are configured by default to use TFTP protocol. If IP phones are configured to obtain configurations from the TFTP server, they will connect to the TFTP server and download the configuration file(s) during startup.
  • Page 30 If the IP phone has successfully passed through these steps, it starts up properly and is ready for use. Icons associated with different features may appear on the LCD screen. The following table provides a description for each icon on SIP-T4X IP phone models. T48G T46G...
  • Page 31 Getting Started T48G T46G T42G/T41P Description Hands-free speakerphone mode Handset mode Headset mode Voice Mail Text Message Auto Answer Do Not Disturb Call Forward Call Hold Call Mute Ringer volume is 0 Phone Lock Multi-lingual lowercase letters input mode Multi-lingual uppercase letters input mode Alphanumeric input mode Numeric input mode...

  • Page 32: Phone User Interface

    Administrator’s Guide for SIP-T4X IP Phones T48G T46G T42G/T41P Description Recording box is full A call cannot be recorded Recording starts successfully Recording cannot be started Recording cannot be stopped VPN is enabled Bluetooth mode is on Bluetooth headset is both paired...
  • Page 33 Specific features access is restricted to the administrator. These specific features are password protected by default. The default password is “admin“(case-sensitive). Not all features are available on phone user interface. For more information, refer to Yealink phone-specific user guide, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 34 Administrator’s Guide for SIP-T4X IP Phones file is named after the MAC address of the IP phone. This file is stored locally on the IP phone and can also be uploaded to the provisioning server. The MAC-local CFG file enables the phone to protect personalized settings. For more...
  • Page 35 Getting Started downloading resource files and upgrading firmware. The transfer protocol is used to download files from the provisioning server. IP phones support several transport protocols for provisioning, including FTP , TFTP , HTTP , and HTTPS protocols. And you can specify the transport protocol in the provisioning server address, for example, http://xxxxxxx.

  • Page 36: Web User Interface

    MAC-oriented configuration file will override the same one in the common configuration file. Yealink supplies configuration files for each phone model, which is delivered with the IP phone firmware. The configuration files, supplied with each firmware release, must be used with that release.
  • Page 37 For more information on the above methods, refer to Yealink_SIP-T2_Series_T4_Series_IP_Phones_Auto_Provisioning_Guide , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. In order to get your IP phones running, you must perform basic network setup, such as IP address and subnet mask configuration. This section describes how to configure basic network parameters for IP phones.
  • Page 38 Administrator’s Guide for SIP-T4X IP Phones Parameter DHCP Option Description Specify a list of time servers available to the Time Server client. Domain Name Specify a list of domain name servers Server available to the client. Specify a list of MIT-LCS UDP servers Log Server available to the client.
  • Page 39 Getting Started Procedure DHCP can be configured using the configuration files or locally. Configure DHCP on the IP phone. Parameter: network.internet_port.type Configure static DNS address <MAC>.cfg when DHCP is used. Parameters: Configuration File network.primary_dns network.secondary_dns Configure the IP phone to use manually configured static IPv4 DNS.
  • Page 40 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default effect. Web User Interface: Network->Basic->IPv4 Config Phone User Interface: Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4->Type network.static_dns_enable 0 or 1 Description: Enables or disables the IP phone to use manually configured static IPv4 DNS when the Internet (WAN) port type for IPv4 is configured as DHCP .
  • Page 41 Getting Started Parameters Permitted Values Default Description: Configures the secondary IPv4 DNS server when the static IPv4 DNS is enabled. Example: network.secondary_dns = 202.101.103.54 Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 42 Administrator’s Guide for SIP-T4X IP Phones 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 43 Getting Started Subnet Mask  Default Gateway  Primary DNS  Secondary DNS  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...
  • Page 44 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Web User Interface: Network->Basic->IPv4 Config Phone User Interface: Menu->Advanced (default password: admin) ->Network->WAN Port->IPv4->Type network.ip_address_mode 0, 1 or 2 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 45 Getting Started Parameters Permitted Values Default Description: Configures the IPv4 subnet mask when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP Address. Example: network.internet_port.mask = 255.255.255.0 Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 46 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the primary IPv4 DNS server when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP Address.
  • Page 47 Getting Started Select the 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 reboot. Click OK to reboot the phone. To configure a static IPv4 address via web user interface: Click on Network->Basic.
  • Page 48 Administrator’s Guide for SIP-T4X IP Phones To configure the IP address mode via phone user interface: Press Menu->Advanced (default password: admin) ->Network->WAN Port. Press to highlight the IP Address Mode field. Press to select IPv4 or IPv4&IPv6 from the IP Address Mode field.
  • Page 49 Getting Started 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&IPv6.
  • Page 50 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the password for PPPoE connection when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet port type is configured as PPPoE. Example: network.pppoe.password = yealink123 Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 51 Getting Started Press the Save soft key to accept the change. The LCD screen will prompt “Reboot now?”. Press OK soft key to reboot the phone and make settings effective or the Cancel soft key to cancel. Two Ethernet ports on the back of the IP phone: Internet port and PC port. Three optional methods of transmission configuration for IP phone Internet or PC Ethernet ports: Auto-negotiation ...
  • Page 52 Administrator’s Guide for SIP-T4X IP Phones Full-duplex Full-duplex transmission refers to transmitting voice or data in both directions at the same time; this means one device can send data on the line while receiving data. You can configure the full-duplex transmission on both Internet port and PC port for IP phones to transmit in 10Mbps, 100Mbps or 1000Mbps (1000Mbps is not applicable to SIP-T41P).
  • Page 53 Getting Started Parameters Permitted Values Default Description: Configures the transmission method and speed of the Internet (WAN) port. 0-Auto negotiate 1-Full duplex, 10Mbps 2-Full duplex, 100Mbps 3-Half duplex, 10Mbps 4-Half duplex, 100Mbps 5-Full duplex, 1000Mbps (not applicable to SIP-T41P) Note: We recommend that you do not change this parameter. If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 54 Click Confirm to accept the change. The PC port on the back of the IP phone is used to connect a PC. You can enable or disable the PC (LAN) port on SIP-T4X IP phones via web user interface or using configuration files.
  • Page 55 Getting Started Parameter Permitted Values Default Description: Enables or disables the PC (LAN) port. 0-Disabled 1-Auto Negotiation Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->PC Port->PC Port Active Phone User Interface: None To enable the PC port via web user interface:...
  • Page 56 Administrator’s Guide for SIP-T4X 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. This section provides information on upgrading the IP phone firmware. Two methods of...
  • Page 57 Getting Started A dialog box pops up to prompt “Firmware of the SIP phone will be updated. It will take 5 minutes to complete. Please don't power off!”. Click OK to confirm the upgrade. Note Do not close and refresh the browser when the IP phone is upgrading firmware via web user interface.
  • Page 58 Administrator’s Guide for SIP-T4X IP Phones auto_provision.weekly.end_time auto_provision.weekly.dayofweek Specify the access URL of firmware. Parameter: firmware.url Configure the way for the IP phone to check for configuration files. Local Web User Interface Navigate to: http://<phoneIPAddress>/servlet?p =settings-autop&q=load Details of Configuration Parameters:...
  • Page 59 Getting Started Parameters Permitted Values Default auto_provision.repeat.minutes Integer from 1 to 43200 1440 Description: Configures the interval (in minutes) for the IP phone to perform an auto provisioning process repeatedly. Note: It works only if the value of the parameter “auto_provision.repeat.enable” is set to 1(Enabled).
  • Page 60 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the end time of the day for the IP phone to perform an auto provisioning process weekly Note: It works only if the value of the parameter “auto_provision.weekly.enable” is set to 1(Enabled).
  • Page 61 Getting Started Parameters Permitted Values Default Description: Configures the access URL of the firmware file. Example: firmware.url = http://192.168.1.20/2.73.0.40.rom Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Settings->Upgrade->Select and Upgrade Firmware Phone User Interface: None To configure the way for the IP phone to check for new configuration files via web user...
  • Page 62 Administrator’s Guide for SIP-T4X IP Phones When the “Power On” is set to On, the IP phone will check configuration files stored on the provisioning server during startup and then will download firmware from the server.
  • Page 63 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 64 Administrator’s Guide for SIP-T4X IP Phones Anonymous Call  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 ...
  • Page 65 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 66 0-Disabled (power indicator LED is off) 1-Enabled (power indicator LED is solid red) Note: The old parameter “features.power_led_on” is also applicable to IP phones and “features.idle_talk_power_led_flash_enable” is also applicable to SIP-T4X IP phones. Web User Interface: Features->Power LED->Common Power Light On...
  • Page 67 Configuring Basic Features Permitted Parameters Default Values Web User Interface: Features->Power LED->Voice/Text Mail Power Light Flash Phone User Interface: None phone_setting.mute_power_led_flash_enable 0 or 1 Description: Enables or disables the power indicator LED to flash when a call is mute. 0-Disabled (power indicator LED does not flash) 1-Enabled (power indicator LED fast flashes (300ms) red) Web User Interface: Features->Power LED->Mute Power Light Flash...
  • Page 68 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameters Default Values None To configure the power Indicator LED via web user interface: Click on Features->Power LED. Select the desired value from the pull-down list of Common Power Light On. Select the desired value from the pull-down list of Ringing Power Light Flash.
  • Page 69 Configuring Basic Features Procedure Notification popups can be configured using the configuration files or locally. Configure notification popups. Parameters: features.voice_mail_popup.enable <y0000000000xx>.cf Configuration File features.missed_call_popup.enable features.forward_call_popup.enable features.text_message_popup.enable Configure notification popups. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet?p=f eatures-notifypop&q=load Details of Configuration Parameters: Parameters Permitted Values Default...
  • Page 70 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Note: It is only applicable to IP phones running firmware version 73 or later. Web User Interface: Features->Notification Popups->Display Missed Call Popup Phone User Interface: None features.forward_call_popup.enable 0 or 1...
  • Page 71 Configuring Basic Features Select the desired value from the pull-down list of Display Text Message Popup. Click Confirm to accept the change. Contrast determines the readability of the texts displayed on the LCD screen. Adjusting the contrast to a comfortable level can optimize the screen viewing experience. When configured properly, contrast allows users to read the LCD’s display with minimal eyestrain.
  • Page 72 Administrator’s Guide for SIP-T4X IP Phones Details of the Configuration Parameter: Parameter Permitted Values Default phone_setting.contrast Integer from 1 to 10 Description: Configures the LCD’s contrast of EXP40 connected to SIP-T48G/T46G IP phones. Note: We recommend that you set the contrast of the LCD screen to 6 as a more comfortable level.
  • Page 73 Configuring Basic Features Procedure Wallpaper can be configured using the configuration files or locally. Configures the wallpaper displayed on the IP phone. Parameter: phone_setting.backgrounds Configuration File <y0000000000xx>.cfg Specify the access URL of the custom wallpaper. Parameter: wallpaper_upload.url Upload the custom wallpaper.
  • Page 74 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Web User Interface: Settings->Preference->Wallpaper Phone User Interface: Menu->Basic->Display-> Wallpaper URL within 511 wallpaper_upload.url Blank characters Description: Configures the access URL of the wallpaper image. Example: wallpaper_upload.url = http://192.168.10.25/wallpaper.jpg Note: It is only applicable to SIP-T48G/T46G IP phones.
  • Page 75 IP phone is inactive. Backlight turns off quickly if a short backlight time is configured, this may not give users enough time to read messages. Backlight time is applicable to SIP-T4X IP phones and EXP40 connected to SIP-T48G/T46G IP phones.
  • Page 76 Administrator’s Guide for SIP-T4X IP Phones SIP-T48G and SIP-T46G IP phones and the connected EXP40. Backlight Inactive Level is only applicable to SIP-T48G and SIP-T46G IP phones. Procedure Backlight can be configured using the configuration files or locally. Configure the backlight of the LCD screen.
  • Page 77 Configuring Basic Features Parameters Permitted Values Default 1-Low Note: It is only applicable to SIP-T48G/T46G IP phones. Web User Interface: Settings->Preference->Backlight Inactive Level Phone User Interface: Menu->Basic->Display->Backlight->Backlight Inactive Level 0, 15, 30, 60, 120, 300, 600 or phone_setting.backlight_time 1800 Description: Configures the delay time (in seconds) to change the intensity of the LCD screen when the IP phone is inactive.
  • Page 78 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Backlight Time (seconds). Click Confirm to accept the change. To configure the backlight via phone user interface: Press Menu->Basic->Display->Backlight. Press , or the Switch soft key to select the desired level from the Backlight Active Level field.
  • Page 79 Configuring Basic Features The following figure shows an incoming call from Yealink 4604 to Yealink 4603 on the SIP-T46G IP phone. You can customize the call information to be displayed on the IP phone as required. IP phones support five call information display methods: Number+Name, Name, Name+Number, Number and Full Contact Info (display name<sip:xxx@domain.com>).
  • Page 80 Administrator’s Guide for SIP-T4X IP Phones Details of Configuration Parameters: Parameters Permitted Values Default phone_setting.contact_photo_display.enable 0 or 1 Description: Enables or disables the IP phone to display contact avatar when it receives an incoming call, dials an outgoing call or engages in a call.
  • Page 81 Configuring Basic Features Parameters Permitted Values Default Web User Interface: Settings->Call Display->Call Information Display Method Phone User Interface: None To configure call display features via web user interface: Click on Settings->Call Display. Select the desired value from the pull-down list of Display Contact Photo. Select the desired value from the pull-down list of Display Called Party Information.
  • Page 82 Administrator’s Guide for SIP-T4X IP Phones network.port.https Specify the web access type, HTTP port and HTTPS port. Web User Interface Navigate to: Local http://<phoneIPAddress>/servl et?p=network-adv&q=load Specify the web access type, Phone User Interface HTTP port and HTTPS port. Details of Configuration Parameters:...
  • Page 83 Configuring Basic Features Parameters Permitted Values Default Port 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 84 Administrator’s Guide for SIP-T4X IP Phones The default HTTPS port is 443. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the phone. To configure the web server type via phone user interface: Press Menu->Advanced (default password: admin) ->Network->Webserver Type.
  • Page 85 Configuring Basic Features user 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.
  • Page 86 Administrator’s Guide for SIP-T4X 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 IP phone with the user credential, the user needs to enter the current user password in the Old Password field.
  • Page 87 Configuring Basic Features Details of the Configuration Parameter: Parameter Permitted Values Default security.user_password String within 32 characters admin Description: Configures the password of the administrator for web server access. The IP phone uses “admin” as the default administrator password. Example: security.user_password = admin:password123 means setting the password of administrator (current user name is “admin”) to password123.
  • Page 88 Administrator’s Guide for SIP-T4X IP Phones Press the Save soft key to accept the change. Phone lock is used to lock the IP phone to prevent it from unauthorized use. Phone lock is not applicable to SIP-T48G IP phones. Once the IP phone is locked, a user must enter the password to unlock it.
  • Page 89 Configuring Basic Features Navigate to: http://<phoneIPAddress>/servlet?p=dssk ey&model=1&q=load&linepage=1 Configure the phone lock type. Phone User Change the unlock PIN. Interface Assign a phone lock key. Details of Configuration Parameters: Parameters Permitted Values Default phone_setting.phone_lock.enable 0 or 1 Description: Enables or disables phone lock feature. 0-Disabled 1-Enabled Note: It is not applicable to SIP-T48G IP phones.
  • Page 90 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Features->Phone Lock->Phone Lock Type Phone User Interface: Menu->Advanced (default password: admin) ->Phone Settings->Phone Lock->Lock Type characters phone_setting.phone_lock.unlock_pin within 15 digits Description: Configures the password for unlocking the phone. Note: It is not applicable to SIP-T48G IP phones.
  • Page 91 Configuring Basic Features Parameter Permitted Values Default Description: Configures a DSS key as a phone lock key on the IP phone. The digit 50 stands for the key type Phone Lock. For line keys: X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P).
  • Page 92 Administrator’s Guide for SIP-T4X IP Phones Parameter Permitted Values Default When X=13, the default value is 0 ( When X=14, the default value is 2 ( Forward IP phones: SIP-T42G/T41P When X=1, the default value is 28 ( History When X=2, the default value is 61 (...
  • Page 93 Configuring Basic Features Click Confirm to accept the change. To configure a phone lock key via web user interface: Click on DSSKey->Line Key X. In the desired DSS key field, select Phone Lock from the pull-down list of Type. Click Confirm to accept the change. To configure phone lock type via phone user interface: Press Menu->Advanced (default password: admin) ->Phone Settings->Phone Lock.

  • Page 94: Time Zone

    Administrator’s Guide for SIP-T4X IP Phones Press the Save soft key to accept the change. IP phones maintain a local clock and calendar. Time and date are displayed on the idle screen of the IP phone. Time and date are synced automatically from the NTP server by default.
  • Page 95 Configuring Basic Features Option Methods of Configuration Configuration Files Daylight Saving Time Web User Interface Procedure Configuration changes can be performed using the configuration files or locally. Configure NTP by DHCP priority feature and DHCP time features. Parameters: local_time.manual_ntp_srv_prior local_time.dhcp_time Configure the NTP server, time zone and DST.
  • Page 96 Administrator’s Guide for SIP-T4X IP Phones Configure the time and date manually. Configure the time and date formats. Navigate to: http://<phoneIPAddress>/servlet ?p=settings-datetime&q=load Configure the NTP server and time zone. Configure the time and date Phone User Interface manually. Configure the time and date formats.
  • Page 97 Configuring Basic Features Parameters Permitted Values Default Phone User Interface: Menu->Basic->Time & Date->DHCP Time local_time.ntp_server1 IP Address or Domain Name cn.pool.ntp.org Description: Configures the IP address or the domain name of the NTP server 1. Example: local_time.ntp_server1 = 192.168.0.5 Web User Interface: Settings->Time &...
  • Page 98 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default local_time.time_zone -11 to +14 Description: Configures the time zone. For more available time zones, refer to Appendix B: Time Zones on page 511. Example: local_time.time_zone = +8 Web User Interface: Settings->Time &...
  • Page 99 Configuring Basic Features Parameters Permitted Values Default Phone User Interface: Menu->Basic->Time & Date->General->SNTP Setting->Daylight Saving local_time.dst_time_type 0 or 1 Description: Configures the DST time type. 0-By Date 1-By Week Note: It works only if the value of the parameter “local_time.summer_time” is set to 1 (Enabled).
  • Page 100 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Settings->Time & Date->Start Date For DST By Week: Settings->Time & Date->DST Start Month/DST Start Day of Week/DST Start Day of Week Last in Month/ Start Hour of Day Phone User Interface: None local_time.end_time...
  • Page 101 Configuring Basic Features Parameters Permitted Values Default Note: It works only if the value of the parameter “local_time.summer_time” is set to 1 (Enabled). Web User Interface: Settings->Time & Date->Offset(minutes) Phone User Interface: None local_time.manual_time_enable 0 or 1 Description: Configures the IP phone to obtain time from the NTP server or manual settings. 0-NTP 1-Manual Web User Interface:...
  • Page 102 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the date format. 0-WWW MMM DD 1-DD-MMM-YY 2-YYYY-MM-DD 3-DD/MM/YYYY 4-MM/DD/YY 5-DD MMM YYYY 6-WWW DD MMM Web User Interface: Settings->Time & Date->Date Format Phone User Interface: Menu->Basic->Time & Date->Time & Date Format->Date Format To configure NTP by DHCP priority feature via web user interface: Click on Settings->Time &...
  • Page 103 Configuring Basic Features 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. If you select Enabled, do one of the following: Mark the DST By Date radio box in the Fixed Type field.
  • Page 104 Administrator’s Guide for SIP-T4X IP Phones Enter the desired time in the End Hour of Day field. Enter the desired offset time in the Offset (minutes) field. Click Confirm to accept the change. To configure the time and date manually via web user interface: Click on Settings->Time &...
  • Page 105 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->Basic->Time & Date->General->SNTP Setting. Press , or the Switch soft key to select the time zone that applies to your area from the Time Zone field.
  • Page 106 Administrator’s Guide for SIP-T4X IP Phones IP phones support multiple languages. Languages used on the phone user interface and web user interface can be specified respectively as required. The following table lists the languages supported by the phone user interface and the web user interface respectively.
  • Page 107 Configuring Basic Features Associated Language Pack Available Language for SIP-T48G/T46G/T42G/T41P 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 108 Administrator’s Guide for SIP-T4X IP Phones The following shows a portion of the language pack “000.GUI.English.lang” for the phone user interface. The following table lists available languages, associated language packs and note language packs for the web user interface: Associated Note...
  • Page 109 Configuring Basic Features When adding a new language pack for the web user interface, the language pack must be formatted as “Y. name.js” (Y starts from 12, “name” is replaced with the language name). If the language name is the same as the existing one, the existing language file will be overridden by the new uploaded one.
  • Page 110 Administrator’s Guide for SIP-T4X IP Phones The following shows a portion of the note language pack “1.English_note.xml” for the web user interface: Note If the characters in the custom language pack are not supported by the phone, the IP phone will display “?” instead.
  • Page 111 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 112 Administrator’s Guide for SIP-T4X IP Phones Parameter Permitted Values Default 010.GUI.Russian.lang). gui_lang.delete = http://localhost/010.GUI.Russian.lang Web User Interface: None Phone User Interface: None wui_lang.url URL within 511 characters Blank Description: Configures the access URL of the language pack for the web user interface.
  • Page 113 Configuring Basic Features Parameter Permitted Values Default Web User Interface: None Phone User Interface: None http://localhost/all or wui_lang.delete Blank name http://localhost/Y. Description: Deletes customized language packs and note language packs of the web user interface. Example: Delete all customized language packs and note language packs of the web user interface.
  • Page 114 Administrator’s Guide for SIP-T4X 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...
  • Page 115 Input method customization allows users to customize the existing input method on IP phones. You can first customize the Yealink-supplied input method file “ime.txt”, and then download it to the IP phone. IP phones support 5 input methods: 2aB, abc, Abc, 123,...
  • Page 116 Administrator’s Guide for SIP-T4X IP Phones The following shows a portion of the input method file “ime.txt”:...
  • Page 117 Configuring Basic Features You can add new characters or adjust the character order of the existing input method. The following show an example of adding the Russian characters for the input method “abc”. Note When adding new characters for the existing input method, ensure that the added characters are supported by IP phones.
  • Page 118 Administrator’s Guide for SIP-T4X IP Phones Specify the default input method when searching for contacts. Parameter: directory.search_default_input_m ethod Details of Configuration Parameters: Parameters Permitted Values Default gui_input_method.url URL within 511 characters Blank Description: Configures the access URL of the custom input method file.
  • Page 119 Configuring Basic Features Parameters Permitted Values Default Description: Specify the default input method for searching the contacts. Example: directory.search_default_input_method = abc Web User Interface: None Phone User Interface: None Logo customization allows unifying the IP phone appearance or displaying a custom image on the idle screen such as a company logo, instead of the default system logo.
  • Page 120 Administrator’s Guide for SIP-T4X IP Phones Details of Configuration Parameters: Parameters Permitted Values Default phone_setting.lcd_logo.mode 0, 1 or 2 Description: Configures the logo mode of the LCD screen. 0-Disabled 1-System logo 2-Custom logo If it is set to 0 (Disabled), the IP phone is not allowed to display a logo.
  • Page 121 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 custom logo screen and the idle screen are displayed alternately. Softkey layout is used to customize the soft keys at the bottom of the LCD screen to best meet users’...
  • Page 122 Administrator’s Guide for SIP-T4X IP Phones Call State Default Soft Key Optional Soft Key Answer Empty Forward Switch CallIn Silence Reject Empty Empty Empty Switch Connecting Empty Cancel Connecting Transfer Empty Empty Switch SemiAttendTrans Empty Cancel Send Empty History Delete...
  • Page 123 Configuring Basic Features Call State Default Soft Key Optional Soft Key Reject PriHold Park GPark Transfer Empty Resume Switch Hold NewCall Answer Cancel Reject Empty Empty Empty Switch Held Empty Answer Cancel Reject NewCall Transfer Empty PreTrans (not Directory applicable to Delete Switch SIP-T48G)
  • Page 124 Administrator’s Guide for SIP-T4X 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.
  • Page 125 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 126 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default None 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.
  • Page 127 Configuring Basic Features The selected soft key appears in the Selected Softkeys column. If more than four soft keys are selected, a More soft key appears on the LCD screen, and the selected soft keys are displayed in two pages. Repeat the step 4 to add more soft keys to the Selected Softkeys column.
  • Page 128 Administrator’s Guide for SIP-T4X IP Phones Configure a send key. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Web User Interface Configure a key tone and send tone. Local Navigate to: http://<phoneIPAddress>/servlet ?p=features-audio&q=load Configure the send key. Phone User Interface Configure a key tone.
  • Page 129 Configuring Basic Features Parameters Permitted Values Default If it is set to 1 (Enabled), the IP phone will play a tone when a user presses a key on your phone phone. Web User Interface: Features->Audio->Key Tone Phone User Interface: Menu->Basic->Sound->Key Tone features.send_key_tone 0 or 1 Description:...
  • Page 130 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Key As Send. Click Confirm to accept the change. To configure a key tone and send tone via web user interface: Click on Features->Audio. Select the desired value from the pull-down list of Key Tone.
  • Page 131 Configuring Basic Features To configure a key tone via web user interface: Press Menu->Basic->Sound->Key Tone. 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.
  • Page 132 Administrator’s Guide for SIP-T4X IP Phones The square bracket "[]" can be used as a placeholder for a single character which matches any of a set of characters. Example: "91[5-7]1234" would match “9151234”, “9161234”, “9171234”. The parenthesis "( )" can be used to group together patterns, for instance, to logically combine two or more patterns.
  • Page 133 Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default dialplan.replace.prefix.X String within 32 characters Blank (X ranges from 1 to 100) Description: Configures the entered number to be replaced. Example: dialplan.replace.prefix.1 = 00 Web User Interface: Settings->Dial Plan->Replace Rule->Prefix Phone User Interface: None dialplan.replace.replace.X...
  • Page 134 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default 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 135 Configuring Basic Features Dial-now is a string used to match the numbers entered by the user. When entered numbers match the predefined dial-now rule, IP phones will automatically dial out the numbers without pressing the send key. IP phones support up to 100 dial-now rules, which can be created either one by one or in batch using a dial-now rule template.
  • Page 136 Administrator’s Guide for SIP-T4X IP Phones Details of Configuration Parameters: Parameters Permitted Values Default dialplan.dialnow.rule.X String within 511 characters Blank (X ranges from 1 to 100) Description: Configures the dial-now rule (the string used to match the numbers entered by the user).
  • Page 137 Configuring Basic Features Parameters Permitted Values Default Description: Configures the delay time (in seconds) for the dial-now rule. When entered numbers match the predefined dial-now rule, the IP phone will automatically dial out the entered number after the specified delay time. Web User Interface: Features->General Information->Time-Out for Dial-Now Rule Phone User Interface:...
  • Page 138 Administrator’s Guide for SIP-T4X IP Phones If you leave the field blank or enter 0, the dial-now rule will apply to all accounts on the IP phone. Click Add to add the dial-now rule. To configure the delay time for the dial-now rule via web user interface: Click on Features->General Information.
  • Page 139 Configuring Basic Features Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country. When the entered numbers match the predefined area code rule, the IP phone will automatically add the area code before the numbers when dialing out them.
  • Page 140 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Web User Interface: Settings->Dial Plan->Area Code->Code Phone User Interface: None dialplan.area_code.min_len Integer from 1 to 15 Description: Configures the minimum length of the entered numbers. Web User Interface: Settings->Dial Plan->Area Code->Min Length (1-15)
  • Page 141 Configuring Basic Features Parameters Permitted Values Default None To configure an area code rule via web user interface: Click on Settings->Dial Plan->Area Code. Enter desired values in the Code, Min Length (1-15) and Max Length (1-15) fields. Enter the desired line ID in the Account field or leave it blank. If you leave the field blank or enter 0, the area code rule will apply to all accounts on the IP phone.
  • Page 142 Administrator’s Guide for SIP-T4X IP Phones ?p=settings-blackout&q=load Details of Configuration Parameters: Parameters Permitted Values Default dialplan.block_out.number.X String within 32 characters Blank (X ranges from 1 to 10) Description: Configures the block out numbers. Example: dialplan.block_out.number.1 = 6666 Web User Interface: Settings->Dial Plan->Block Out->BlockOut NumberX...
  • Page 143 Configuring Basic Features If you leave the field blank or enter 0, the block out rule will apply to all accounts on the IP phone. Click Confirm to add the block out rule. Hotline is a point-to-point communication link in which a call is automatically directed to the preset hotline number.
  • Page 144 Administrator’s Guide for SIP-T4X IP Phones ?p=features-general&q=load Configure the hotline number. Specify the time (in seconds) the Phone User Interface IP phone waits to automatically dial out the hotline number. Details of Configuration Parameters: Parameter Permitted Values Default features.hotline_number String within 32 characters...
  • Page 145 Configuring Basic Features 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->Others->Hot Line. Enter the hotline number in the Number field. Enter the delay time in the HotLine Delay field. Press the Save soft key to accept the change.
  • Page 146 Administrator’s Guide for SIP-T4X IP Phones Procedure Off hook hot line dialing can be configured using the configuration files. Configure off hook hot line dialing feature. Parameter: account.X.auto_dial_enable Configuration File <y0000000000xx>.cfg Specify the number that the phone first dials out.
  • Page 147 Configuring Basic Features Parameter Permitted Values Default account X. X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P). Note: It works only if the value of the parameter “account.X.auto_dial_enable” is set to 1 (Enabled).
  • Page 148 Administrator’s Guide for SIP-T4X IP Phones Parameter Permitted Values Default Description: Configures the access URL of the directory template. Example: directory_setting.url = http://192.168.1.20/favorite_setting.xml Web User Interface: Directory->Setting->Directory Phone User Interface: None To configure the directory via web user interface: Click on Directory->Setting.
  • Page 149 Configuring Basic Features Search source list in dialing allows the IP phone to automatically search entries from the search source list based on the entered string, and display results on the pre-dialing screen. The search source list can be Local Directory, History, Remote Phone Book and LDAP .
  • Page 150 Administrator’s Guide for SIP-T4X IP Phones To remove a list from the Enabled column, select the desired list and then click To adjust the display order of the enabled list, select the desired list, and click Click Confirm to accept the change.
  • Page 151 Configuring Basic Features 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 placed calls, received calls, missed calls and the forwarded calls in the call log lists.
  • Page 152 Administrator’s Guide for SIP-T4X IP Phones Press , or the Switch soft key to select the desired value from the History Record field. Press the Save soft key to accept the change. Missed call log allows IP phones to display the number of the missed calls with an indicator icon on the idle screen, and to log the missed calls in the Missed Calls list when the IP phone misses calls.
  • Page 153 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...
  • Page 154 Administrator’s Guide for SIP-T4X IP Phones 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.
  • Page 155 Configuring Basic Features Select the desired group 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. Enter the name and the office, mobile or other numbers in the corresponding fields. Select the desired ring tone from the pull-down list of Ring Tone.
  • Page 156 Administrator’s Guide for SIP-T4X IP Phones It is not applicable to SIP-T42G and SIP-T41P IP phones. Click Add to add the contact. To import an XML contact list file via web user interface: Click on Directory->Local Directory. Click Browse to locate a contact list file (the file format must be *.xml) from your local system.
  • Page 157 Configuring Basic Features The web user interface prompts "The original contact will be covered, Continue?". Click OK to complete importing the contact list. To import a CSV contact list file via web user interface: Click on Directory->Local Directory. Click Browse to locate a contact list file (the file format must be *.csv) from your local system.
  • Page 158 Administrator’s Guide for SIP-T4X IP Phones Press to select the desired group ring tone from the Ring Tones field. Press the Save soft key to accept the change or the Back soft key to cancel. To add a contact to the local directory via phone user interface: Press Menu->Directory->Local Directory.
  • Page 159 Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default phone_setting.predial_autodial 0 or 1 Description: Enables or disables live dialpad feature. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will automatically dial out the entered phone number in the pre-dialing screen without pressing a send key.
  • Page 160 Administrator’s Guide for SIP-T4X IP Phones 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 161 Configuring Basic Features ?p=features-general&q=load Phone User Interface Configure call waiting. Details of Configuration Parameters: Parameters Permitted Values Default call_waiting.enable 0 or 1 Description: Enables or disables call waiting feature. 0-Disabled 1-Enabled If it is set to 0 (Disabled), a new incoming call is automatically rejected by the IP phone with a busy message while during a call.
  • Page 162 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Configures the call waiting off code to deactivate the server-side call waiting feature. The IP phone will send the call waiting off code to the server when you deactivate call waiting feature on the IP phone.
  • Page 163 Configuring Basic Features Select the desired value from the pull-down list of Call Waiting Tone. Click Confirm to accept the change. To configure call waiting and call waiting tone via phone user interface: Press Menu->Features->Call Waiting. Press , or the Switch soft key to select the desired value from the Call Waiting field.
  • Page 164 Administrator’s Guide for SIP-T4X IP Phones Details of Configuration Parameters: Parameters Permitted Values Default auto_redial.enable 0 or 1 Description: Enables or disables the IP phone to automatically redial the dialed number when the callee is temporarily unavailable. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will dial the previous dialed out number automatically when the dialed number is temporarily unavailable.
  • Page 165 Configuring Basic Features To configure auto redial via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Auto Redial. Enter the desired time interval (in seconds) in the Auto Redial Interval (1~300s) field. The default waiting time is 10s.
  • Page 166 Administrator’s Guide for SIP-T4X IP Phones Procedure Auto answer can be configured using the configuration files or locally. Configure auto answer. <MAC>.cfg Parameter: account.X.auto_answer Configuration File Specify a period of delay time for auto answer. <y0000000000xx>.cfg Parameter: features.auto_answer_delay Configure auto answer.
  • Page 167 Configuring Basic Features Parameters Permitted Values Default Phone User Interface: Menu->Features ->Auto Answer->Line X->Auto Answer features.auto_answer_delay Integer from 1 to 4 (X ranges from 1 to 16) Description: Configures the delay time (in seconds) before the IP phone automatically answers an incoming call.
  • Page 168 Administrator’s Guide for SIP-T4X IP Phones Enter the desired time (in seconds) in the Auto-Answer Delay (1~4s) field. Click Confirm to accept the change. To configure auto answer via phone user interface: Press Menu->Features->Auto Answer. Select the desired line and then press the Enter soft key.
  • Page 169 Configuring Basic Features Procedure Call completion can be configured using the configuration files or locally. Configure call completion. Configuration File <y0000000000xx>.cfg Parameter: features.call_completion_enable Configure call completion. Navigate to: Web User Interface http://<phoneIPAddress>/servlet Local ?p=features-general&q=load Phone User Interface Configure call completion. Details of the Configuration Parameter: Parameter Permitted Values...
  • Page 170 Administrator’s Guide for SIP-T4X IP Phones 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->Others->Call Completion. Press , or the Switch soft key to select the desired value from the Call Completion field.
  • Page 171 Configuring Basic Features User-Agent: Yealink SIP-T46G 28.73.0.10 Privacy: id Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync P-Preferred-Identity: <sip:1012@10.2.1.199> Content-Length: 302 The anonymous call on code and anonymous call off code configured on IP phones are used to activate/deactivate the server-side anonymous call feature. They may vary on different servers.
  • Page 172 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P). Web User Interface: Account->Basic->Local Anonymous Phone User Interface: Menu->Features->Anonymous->Line X ->Local Anonymous...
  • Page 173 Configuring Basic Features Parameters Permitted Values Default Example: account.1.anonymous_call_oncode = *72 Note: It works only if the value of the parameter “account.X.send_anonymous_code” is set to 1 (On Code). Web User Interface: Account->Basic->Send Anonymous Code->On Code Phone User Interface: Menu->Features->Anonymous->Line X ->On Code account.X.anonymous_call_offcode String within 32 characters Blank...
  • Page 174 Administrator’s Guide for SIP-T4X IP Phones (Optional.) Enter the anonymous call off code in the Off Code field. Click Confirm to accept the change. To configure the anonymous call via phone user interface: Press Menu->Features->Anonymous. Select the desired line and then press Enter soft key.
  • Page 175 Configuring Basic Features Procedure Anonymous call rejection can be configured using the configuration files or locally. Configure anonymous call rejection. Parameters: account.X.reject_anonymous_call Configuration File <MAC>.cfg account.X.send_anonymous_rejection _code account.X.anonymous_reject_oncode account.X.anonymous_reject_offcode Configure anonymous call rejection. Navigate to: Web User Interface http://<phoneIPAddress>/servlet?p=a Local ccount-basic&q=load&acc=0 Phone User Interface Configure anonymous call rejection.
  • Page 176 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the IP phone to send anonymous rejection on/off code to activate/deactivate the server-side anonymous rejection feature for account X. 0-Off Code 1-On Code If it is set to 0 (Off Code), the IP phone will send reject off code to deactivate the server-side anonymous rejection feature when you enable or disable anonymous rejection feature on the phone.
  • Page 177 Configuring Basic Features Parameters Permitted Values Default account.X.anonymous_reject_offcode String within 32 characters Blank Description: Configures the anonymous call rejection off code to deactivate the server-side anonymous call rejection feature for account X. The IP phone will send anonymous call rejection code to the server when you deactivate anonymous call rejection feature on the IP phone.
  • Page 178 Administrator’s Guide for SIP-T4X IP Phones (Optional.) Enter the anonymous call rejection off code in the Off Code field. Click Confirm to accept the change. To configure anonymous call rejection via phone user interface: Press Menu->Features->Anonymous. Select the desired line and then press Enter soft key.
  • Page 179 Configuring Basic Features Return Message When DND This feature defines the return code and the reason of the SIP response message for the rejected incoming call when DND is enabled on IP phones. The caller’s LCD screen displays the received return code. Procedure DND can be configured using the configuration files or locally.
  • Page 180 Administrator’s Guide for SIP-T4X IP Phones p=features-forward&q=load Specify return code and reason of the SIP response message. Navigate to: http://<phoneIPAddress>/servlet? p=features-general&q=load Assign a DND key. Phone User Interface Configure DND. Details of Configuration Parameters: Parameters Permitted Values Default account.X.dnd.enable 0 or 1...
  • Page 181 Configuring Basic Features Parameters Permitted Values Default Web User Interface: Features->Forward& DND->DND->DND On Code Phone User Interface: Menu->Features->DND->Account->On Code account.X.dnd.off_code String within 32 characters Blank Description: Configures the DND off code to deactivate the server-side DND feature for account X when the DND mode is configured as Custom.
  • Page 182 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Enables or disables DND feature when the DND mode is configured as Phone. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will reject incoming calls on all accounts.
  • Page 183 Configuring Basic Features Parameters Permitted Values Default Description: Configures a return code and reason of SIP response messages when rejecting an incoming call by DND. A specific reason is displayed on the caller’s phone LCD screen. 404-No Found 480-Temporarily Unavailable 486-Busy Here If it is set to 486 (Busy Here), the caller’s LCD screen will display the reason “Busy Here”...
  • Page 184 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameter Default Values For line keys: For SIP-T48G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0. For SIP-T46G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
  • Page 185 Configuring Basic Features Permitted Parameter Default Values When X=7, the default value is 51 ( Switch Account Up When X=8, the default value is 52 ( Switch Account Down When X=9, the default value is 33 (Status). When X=10, the default value is 0 ( When X=13, the default value is 0 ( Web User Interface: / Programable Key...
  • Page 186 Administrator’s Guide for SIP-T4X IP Phones 3) (Optional.) Enter the DND off code in the DND Off Code field. b) If you mark the Custom radio box: 1) Select the desired account from the pull-down list of Account. 2) Mark the desired value in the DND Status field.
  • Page 187 Configuring Basic Features 4) (Optional.) Enter the DND off code in the DND Off Code field. Click Confirm to accept the change. To specify the return code via web user interface: Click on Features->General Information. Select the desired type from the pull-down list of Return Code When DND.
  • Page 188 Administrator’s Guide for SIP-T4X IP Phones 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.
  • Page 189 Configuring Basic Features Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Details of the Configuration Parameter: Parameter Permitted Values Default features.busy_tone_delay 0, 3 or 5 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 190 Administrator’s Guide for SIP-T4X IP Phones 3． Click Confirm to accept the change. Return code when refuse defines the return code and reason of the SIP response message for call rejection. The caller’s LCD screen displays the reason according to the return code received.
  • Page 191 Configuring Basic Features Parameter Permitted Values Default Features->General Information->Return Code When Refuse Phone User Interface: None To specify the return code when refusing a call via web user interface: 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 192 Administrator’s Guide for SIP-T4X IP Phones 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 Configure 180 ring workaround. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=features-general&q=load...
  • Page 193 Configuring Basic Features 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 194 Administrator’s Guide for SIP-T4X IP Phones 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...
  • Page 195 Configuring Basic Features SIP session timers T1, T2 and T4 are SIP transaction layer timers defined in RFC 3261. Timer T1 is an estimate of the Round Trip Time (RTT) of transactions between a SIP client and SIP server. Timer T2 represents the maximum retransmitting time of any SIP request message.
  • Page 196 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default None account.X.advanced.timer_t2 Float from 2 to 40 Description: Configures the session timer T2 (in seconds) for account X. T2 represents the maximum retransmit interval for non-INVITE requests and INVITE responses.
  • Page 197 Configuring Basic Features Enter the desired value in the SIP Session Timer T4 (2.5~60s) field. The default value is 5s. 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 198 Administrator’s Guide for SIP-T4X IP Phones features.play_hold_tone.delay Specify whether RFC 2543 (c=0.0.0.0) outgoing hold signaling is used. Parameters: sip.rfc2543_hold Configure MoH on a per-line basis. <MAC>.cfg Parameter: account.X.music_server_uri Configure the call hold tone and call hold tone delay. Specify whether RFC 2543 (c=0.0.0.0) outgoing hold...
  • Page 199 Configuring Basic Features Parameters Permitted Values Default features.play_hold_tone.delay Integer from 3 to 3600 Description: Configures the interval (in seconds) at which the IP phone plays a hold tone. If it is set to 30 (30s), the IP phone will play a hold tone every 30 seconds when there is a hold call on the IP phone.
  • Page 200 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the address of the Music On Hold server for account X. Examples for valid values: <10.1.3.165>, 10.1.3.165, sip:moh@sip.com, <sip:moh@sip.com>, <yealink.com> or yealink.com. X ranges from 1 to 16 (for SIP-T48G/T46G).
  • Page 201 Configuring Basic Features Select the desired value from the pull-down list of Play Hold Tone. Enter the desired time in the Play Hold Tone Delay field. Click Confirm to accept the change. To configure MoH via web user interface: Click on Account. Select the desired account from the pull-down list of Account.
  • Page 202 Administrator’s Guide for SIP-T4X IP Phones Session timer allows a periodic refresh of SIP sessions through a re-INVITE request, to determine whether a SIP session is still active. Session timer is specified in RFC 4028. IP phones support two refresher modes: UAC and UAS. The UAC mode means refreshing the session from the client, while the UAS mode means refreshing the session from the server.
  • Page 203 Configuring Basic Features Parameters Permitted Values Default Account->Advanced->Session Timer Phone User Interface: None account.X.session_timer.expires Integer from 30 to 7200 1800 Description: Configures the IP phone to refresh the session during a call at regular intervals (in seconds) for account X. If it is set to 1800 (1800s), the IP phone will refresh the session during a call before 1800 seconds.
  • Page 204 Administrator’s Guide for SIP-T4X IP Phones To configure session timer via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced. Select the desired value from the pull-down list of Session Timer.
  • Page 205 Configuring Basic Features Custom: Call forward feature can be configured for each or all accounts.  The server-side call forward settings disable the local call forward settings. If the server-side call forward feature is enabled on any of the IP phone’s registrations, the other registrations are not affected.
  • Page 206 Administrator’s Guide for SIP-T4X IP Phones Configure the call forward mode. Parameter: features.fwd_mode Configure call forward in phone mode. Parameters: forward.always.enable forward.always.target forward.always.on_code forward.always.off_code forward.busy.enable <y0000000000xx>.cf forward.busy.target forward.busy.on_code forward.busy.off_code forward.no_answer.enable forward.no_answer.target forward.no_answer.timeout forward.no_answer.on_code forward.no_answer.off_code features.fwd_diversion_enable Configure forward international. Parameter: forward.international.enable Configure call forward.
  • Page 207 Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default account.X.always_fwd.enable 0 or 1 Description: Enables or disables always forward feature for account X when the call forward mode is configured as Custom 0-Disabled 1-Enabled If it is set to 1 (Enabled), incoming calls to the account X are forwarded to the destination number immediately.
  • Page 208 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default feature for account X . The IP when the call forward mode is configured as Custom phone will send the always forward on code and the pre-configured destination number to the server when you activate always forward feature for account X on the IP phone.
  • Page 209 Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables busy forward feature for account X when the call forward mode is configured as Custom 0-Disabled 1-Enabled If it is set to 1 (Enabled), incoming calls to the account X are forwarded to the destination number when the callee is busy.
  • Page 210 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the busy forward on code to activate the server-side busy forward feature for account X The IP when the call forward mode is configured as Custom. phone will send the busy forward on code and the pre-configured destination number to the server when you activate busy forward feature for account X on the IP phone.
  • Page 211 Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables no answer forward feature for account X when the call forward mode is configured as Custom. 0-Disabled 1-Enabled If it is set to 1 (Enabled), incoming calls to the account X are forwarded to the destination number after a period of ring time.
  • Page 212 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures ring times (N) to wait before forwarding incoming calls for account X when the call forward mode is configured as Custom. Incoming calls will be forwarded when not answered after N*6 seconds.
  • Page 213 Configuring Basic Features Parameters Permitted Values Default X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P). Example: account.1.timeout_fwd.off_code = *77 Web User Interface: Features->Forward& DND->Forward->No Answer Forward ->Off Code Phone User Interface: Menu->Features->Call Forward->Account ->No Answer Forward ->Off Code features.fwd_mode...
  • Page 214 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the destination number the IP phone forwards all incoming calls to. Web User Interface: Features->Forward &DND->Always Forward->Target Phone User Interface: Menu->Features->Call Forward->Always Forward->Forward To forward.always.on_code String within 32 characters...
  • Page 215 Configuring Basic Features Parameters Permitted Values Default 0-Disabled 1-Enabled If it is set to 1 (Enabled), incoming calls are forwarded to the destination number when the callee is busy. Web User Interface: Features->Forward &DND->Forward->Busy Forward->On/Off Phone User Interface: Menu->Features->Call Forward->Busy Forward->Busy Forward forward.busy.target String within 32 characters Blank...
  • Page 216 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Configures the busy forward off code to deactivate the server-side busy forward feature. The IP phone will send the busy forward off code to the server when you deactivate busy forward feature on the IP phone.
  • Page 217 Configuring Basic Features Parameters Permitted Values Default Description: Configures ring times (N) to wait before forwarding incoming calls. Incoming calls will be forwarded when not answered after N*6 seconds. Web User Interface: Features->Forward &DND->Forward->No Answer Forward->After Ring Time (0~120s) Phone User Interface: Menu->Features->Call Forward->No Answer Forward->After Ring Time forward.no_answer.on_code String within 32 characters...
  • Page 218 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Enables or disables the IP phone to present the diversion information when an incoming call is forwarded to your IP phone. 0-Disabled 1-Enabled Web User Interface: Features->General Information->Diversion/History-Info Phone User Interface: None forward.international.enable...
  • Page 219 Configuring Basic Features 4) Select the ring time to wait before forwarding from the pull-down list of After Ring Time (0~120s) (only for the no answer forward). b) If you mark the Custom radio box: 1) Select the desired account from the pull-down list of Account. 2) Mark the desired radio box in the Always Forward/Busy Forward/No Answer Forward field.
  • Page 220 Administrator’s Guide for SIP-T4X IP Phones To configure the forward international feature via web user interface: Click on Features->General Information. 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.
  • Page 221 Configuring Basic Features c) If you select No Answer Forward: 1) Press , or the Switch soft key to select the desired value from the No Answer Forward field. 2) Enter the destination number you want to forward all unanswered incoming calls to in the Forward to field.
  • Page 222 Administrator’s Guide for SIP-T4X IP Phones 1) Press to highlight the Busy Forward field. 2) Press the All Lines soft key. The LCD screen prompts “Copy to all lines?”. 3) Press the OK soft key to accept the change. c) If you select No Answer Forward, you can configure it for a specific account.
  • Page 223 Configuring Basic Features implemented by a REFER method with Replaces in the Refer-To header. Normally, call transfer is completed by pressing the transfer key. Blind transfer on hook and attended transfer on hook features allow the IP phone to complete the transfer through on-hook.
  • Page 224 Administrator’s Guide for SIP-T4X IP Phones Details of Configuration Parameters: Parameters Permitted Values Default transfer.blind_tran_on_hook_enable 0 or 1 Description: Enables or disables the IP phone to complete the blind transfer through on-hook besides pressing the Transfer/Tran soft key or TRAN/TRANSFER key.
  • Page 225 Configuring Basic Features Select the desired values from the pull-down lists of Semi-Attend Transfer, Blind Transfer On Hook and Semi 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 226 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the network conference type for account X. 0-Local Conference 2-Network Conference If it is set to 0 (Local Conference), conferences are set up on the IP phone locally.
  • Page 227 Configuring Basic Features 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 228 Administrator’s Guide for SIP-T4X IP Phones 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:...
  • Page 229 Configuring Basic Features For local conference, all parties drop the call when the conference initiator drops the conference call. For local conference, transfer on conference hang up allows the other two parties to remain connected when the conference initiator drops the conference call.
  • Page 230 Administrator’s Guide for SIP-T4X IP Phones 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.
  • Page 231 Configuring Basic Features Configure directed call pickup features on a phone basis. Parameters: features.pickup.direct_pickup_ enable features.pickup.direct_pickup_c Assign a directed call pickup <y0000000000xx>.cfg key. Parameters: linekey.X.type/ programablekey.X.type linekey.X.line/ programablekey.X.line linekey.X.value/ programablekey.X.value Assign a directed call pickup key. Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&model=1&q=loa d&linepage=1 Configure the directed call pickup feature on a phone...
  • Page 232 Administrator’s Guide for SIP-T4X IP Phones Details of Configuration Parameters: Parameters Permitted Values Default String within 32 characters Blank account.X.direct_pickup_code Description : Configures the directed call pickup code for account X. X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G).
  • Page 233 Configuring Basic Features Parameters Permitted Values Default Description: Configures the directed call pickup code on a phone basis. Example: features.pickup.direct_pickup_code = *97 Note: The directed call pickup code configured on a per-line basis takes precedence over that configured on a phone basis. Web User Interface: Features->Call Pickup->Directed Call Pickup Code Phone User Interface:...
  • Page 234 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default is 0. For SIP-T46G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0. For SIP-T42G IP phones: The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
  • Page 235 Configuring Basic Features Parameters Permitted Values Default When X=10, the default value is 0 ( When X=13, the default value is 0 ( Web User Interface: DSSKey->Line Key X/ Programable Key ->Type Phone User Interface: Menu->Features->DSS Keys->Line Key X->Type 1-6 correspond linekey.X.line/programablekey.X.line Integer from 1 to 16 to the lines 1-6...
  • Page 236 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default For line keys: X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). For programable keys:...
  • Page 237 Configuring Basic Features Enter the directed call pickup code in the Directed Call Pickup Code field. Click Confirm to accept the change. To configure the directed call pickup code on a per-line basis via web user interface: Click on Account. Select the desired account from the pull-down list of Account.
  • Page 238 Administrator’s Guide for SIP-T4X IP Phones Press , or the Switch soft key to 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 directed call pickup code followed by the specific extension in the Value field.
  • Page 239 Configuring Basic Features et?p=dsskey&model=1&q=loa d&linepage=1 Configure the group call pickup feature on a phone basis. Navigate to: http://<phoneIPAddress>/servl et?p=features-callpickup&q=lo Configure the group call pickup code on a per-line basis. Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac Phone User Interface Assign a group call pickup key. Details of Configuration Parameters: Parameters Permitted Values...
  • Page 240 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the group pickup code for account X. X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P).
  • Page 241 Configuring Basic Features Parameters Permitted Values Default Description: Configures a DSS key as a group call pickup key on the IP phone. The digit 23 stands for the key type Group Pickup. For line keys: X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G).
  • Page 242 Administrator’s Guide for SIP-T4X 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 ( When X=12, the default value is 0 (...
  • Page 243 Configuring Basic Features Parameters Permitted Values Default 1 to 12 (for SIP-T42G) 1 to 6 (for SIP-T41P) 1-Line 1 2-Line 2 … 16-Line 16 Web User Interface: DSSKey->Line Key X/ Programable Key->Line Phone User Interface: Menu->Features->DSS Keys->Line Key X->Account ID String within 99 linekey.X.value/ programablekey.X.value blank...
  • Page 244 Administrator’s Guide for SIP-T4X IP Phones Select the desired line from the pull-down list of Line. Click Confirm to accept the change. To configure the group call pickup feature on a phone basis via web user interface: Click on Features->Call Pickup.
  • Page 245 Configuring Basic Features 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 246 Administrator’s Guide for SIP-T4X IP Phones entity="sip:1013@10.2.1.199"> <dialog id="706655206@10.2.8.213" call-id="706655206@10.2.8.213" local-tag="827932784" remote-tag="1887460740" direction="recipient"> <state>early</state> <local> <identity>sip:1013@10.2.1.199</identity> <target uri="sip:1013@10.2.1.199"> </target> </local> <remote> <identity>sip:1011@10.2.1.199</identity> <target uri="sip:1011@10.2.8.213:5063"> </target> </remote> </dialog> </dialog-info> Procedure Dialog-info call pickup can be configured using the configuration files or locally.
  • Page 247 Configuring Basic Features Parameter Permitted Values Default Description: Enables or disables the IP phone to pick up a call according to the SIP header of dialog-info for account X. 0-Disabled 1-Enabled If it is set to 1 (Enabled), call pickup is implemented through SIP signals. X ranges from 1 to 16 (for SIP-T48G/T46G).
  • Page 248 Administrator’s Guide for SIP-T4X IP Phones Recall, also known as last call return, allows users to place a call back to the last caller. Recall is implemented on IP phones using a recall key. Procedure Recall key can be configured using the configuration files or locally.
  • Page 249 Configuring Basic Features Permitted Parameter Default Values Default: For line keys: For SIP-T48G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0. For SIP-T46G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
  • Page 250 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameter Default Values When X=6, the default value is 61 ( Directory When X=7, the default value is 51 ( Switch Account Up When X=8, the default value is 52 ( Switch Account Down When X=9, the default value is 33 (Status).
  • Page 251 Configuring Basic Features Call park allows users to park a call on a special extension and then retrieve it on any other phone in the system. Users can park calls on the extension, known as call park orbit, by pressing a call park key. The current call is placed on hold and can be retrieved on another IP phone.
  • Page 252 Administrator’s Guide for SIP-T4X IP Phones Parameter Permitted Values Default is 0. For SIP-T46G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0. For SIP-T42G IP phones: The default value of the line key 1-12 is 15, and the default value of the line key 13-15 is 0.
  • Page 253 Configuring Basic Features Parameter Permitted Values Default Description: Configures the call park feature code. X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). Example: linekey.2.value = *88 Web User Interface: DSSKey->Line Key X->Line Key X->...
  • Page 254 If the caller has existed in the local directory, the local name assigned to the caller should be preferentially displayed and stored in the call log. 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 255 Configuring Basic Features Parameter Permitted Values Default Description: Configures the presentation of the caller identity when receiving an incoming call for account X. 0-FROM (Derives the name and number of the caller from the “From” header). 1-PAI (Derives the name and number of the caller from the “PAI” header. If the server does not send the “PAI”...
  • Page 256 If the callee has existed 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.
  • Page 257 Configuring Basic Features Procedure COLP can be configured only using the configuration files. Configure the presentation of the callee identity. Configuration File <MAC>.cfg Parameter: account.X.cp_source Details of the Configuration Parameter: Parameter Permitted Values Default account.X.cp_source 0, 1 or 2 Description: Configures the presentation of the callee’s identity for account X.
  • Page 258 Administrator’s Guide for SIP-T4X IP Phones frequency group and the other from a low frequency group. The DTMF keypad is laid out in a 4× 4 matrix, with each row representing a low frequency, and each column representing a high frequency. Pressing a digit key (such as '1') will generate a sinusoidal tone for each of two frequencies (697 and 1209 hertz (Hz)).
  • Page 259 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 260 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the DTMF type for account X. 0-INBAND 1-RFC 2833 2-SIP INFO 3-RFC2833 + SIP INFO If it is set to 0 (INBAND), DTMF digits are transmitted in the voice band.
  • Page 261 Configuring Basic Features Parameters Permitted Values Default Description: Configures the DTMF info type when the DTMF type is configured as “SIP INFO”, “RFC2833 + SIP INFO” for account X. 0-Disabled 1-DTMF-Relay 2-DTMF 3-Telephone-Event X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G).
  • Page 262 Administrator’s Guide for SIP-T4X IP Phones Click on Advanced. Select the desired value from the pull-down list of DTMF Type. If SIP INFO or RFC2833 + SIP INFO is selected, select the desired value from the pull-down list of DTMF Info Type.
  • Page 263 Configuring Basic Features Click Confirm to accept the change. Suppress DTMF display allows IP phones to suppress the display of DTMF digits. The digits are displayed as “*” on the LCD screen. Suppress DTMF display delay defines whether to display the DTMF digits for a short period of time before displaying as “*”. Procedure Configuration changes can be performed using the configuration files or locally.
  • Page 264 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default features.dtmf.hide_delay 0 or 1 Description: Enables or disables the IP phone to display the DTMF digits for a short period before displaying asterisks during an active call. 0-Disabled 1-Enabled Note: It works only if the value of the parameter “features.dtmf.hide” is set to 1 (Enabled).
  • Page 265 Configuring Basic Features Call transfer is implemented via DTMF on some traditional servers. The IP phone sends specified DTMF digits to the server for transferring calls to third parties. Procedure Configuration changes can be performed using the configuration files or locally. Configure transfer via DTMF.
  • Page 266 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: the DTMF digits to be transmitted to perform call transfer. Valid values Configures are: 0-9, *, # and A-D. Example: features.dtmf.transfer = 123 Note: It works only if the value of the parameter “features.dtmf.replace_tran” is set to 1 (Enabled).

  • Page 267: Intercom Key

    Configuring Basic Features Intercom is a useful feature in office environments to quickly connect with an operator or secretary. Users can press an intercom key to automatically initiate an outgoing intercom call with a remote extension. Procedure Intercom key can be configured using the configuration files or locally. Assign an intercom key.
  • Page 268 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0. For SIP-T46G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
  • Page 269 Configuring Basic Features Parameters Permitted Values Default Description: Configures the intercom number. X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). Web User Interface: DSSKey->Line Key X->Line KeyX->Value Phone User Interface: Menu->Features->DSS Keys->Line Key X->Value To configure an intercom key via web user interface:...
  • Page 270 Administrator’s Guide for SIP-T4X IP Phones The IP phone can process incoming calls differently depending on settings. There are four configuration options for incoming intercom calls. Accept Intercom Accept Intercom allows the IP phone to automatically answer an incoming intercom call.
  • Page 271 Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables the IP phone to automatically answer an incoming intercom call. 0-Disabled 1-Enabled If it is set to 0 (Disabled), the IP phone will reject incoming intercom calls and sends a busy signal to the caller.
  • Page 272 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Enables or disables the IP phone to play a warning tone when receiving an intercom call. 0-Disabled 1-Enabled Note: It works only if the value of the parameter “...
  • Page 273 Configuring Basic Features 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 274 Administrator’s Guide for SIP-T4X IP Phones...
  • Page 275 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 276 Administrator’s Guide for SIP-T4X 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 277 Configuring Advanced Features Minimum Nominal Maximum Bellcore Pattern Pattern Cadence Duration Duration Duration Tone (ms) (ms) (ms) Silent Ringing Long 1025 Silent 2975 4000 4400 Ringing Short Silent Ringing Long 1000 1100 Bellcore-dr4 Silent Ringing Short Silent 2975 4000 4400 Bellcore-dr5 Ringing Note...

  • Page 278: Auto Answer

    Administrator’s Guide for SIP-T4X IP Phones Value of N Ring Tone 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 279 Configuring Advanced Features Parameters: features.alert_info_tone distinctive_ring_tones.alert_info .X.text distinctive_ring_tones.alert_info .X.ringer Configure distinctive ring tones feature. Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac Local Web User Interface Configure the internal ringer text and internal ringer file. Navigate to: http://<phoneIPAddress>/servl et?p=settings-ring&q=load Details of Configuration Parameters: Parameters Permitted Values Default account.X.alert_info_url_enable...
  • Page 280 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Enables or disables the IP phone to map the keywords in the Alert-info header to the specified Bellcore ring tones. 0-Disabled 1-Enabled Web User Interface: None Phone User Interface: None distinctive_ring_tones.alert_info.X.text...
  • Page 281 Configuring Advanced Features Parameters Permitted Values Default 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 282 Administrator’s Guide for SIP-T4X IP Phones Select the desired ring tones for each text from the pull-down lists of Internal Ringer File. Click Confirm to accept the change. When receiving a message, the IP phone will play a warning tone. You can customize tones or select specialized tone sets (vary from country to country) to indicate different conditions of the IP phone.
  • Page 283 Configuring Advanced Features Lithuania  India  Italy  Japan  Mexico  New Zealand  Netherlands  Norway  Portugal  Spain  Switzerland  Sweden  Russia  United States  Chile  Czech ETSI  Configured tones can be heard on the IP phone for the following conditions: Condition Description Dial...
  • Page 284 Administrator’s Guide for SIP-T4X IP Phones phone. Parameters: voice.tone.country voice.tone.dial voice.tone.ring voice.tone.busy voice.tone.congestion voice.tone.callwaiting voice.tone.dialrecall voice.tone.info voice.tone.stutter voice.tone.message voice.tone.autoanswer Configure the tones for the IP phone. Local Web User Interface Navigate to: http://<phoneIPAddress>/servl et?p=settings-tones&q=load Details of Configuration Parameters: Parameters Permitted Values Default voice.tone.country...
  • Page 285 Configuring Advanced Features Parameters Permitted Values Default voice.tone.dial String Blank Description: Customizes the dial tone. tonelist = element[,element] [,element]… Where element = [!]Freq1[+Freq2][+Freq3][+Freq4] /Duration Freq: the frequency of the tone (ranges from 200 to 7000 Hz). If it is set to 0Hz, it means the tone is not played.
  • Page 286 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Customizes the tone when the callee is busy. The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”. Note: It works only if the value of the parameter “voice.tone.country” is set to Custom.
  • Page 287 Configuring Advanced Features Parameters Permitted Values Default None voice.tone.dialrecall String Blank Description: Customizes the call back tone. The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”. Note: It works only if the value of the parameter “voice.tone.country” is set to Custom.
  • Page 288 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Custom. Web User Interface: Settings->Tones->Stutter Phone User Interface: None voice.tone.message String Blank Description: Customizes the tone when the IP phone receives a text message or voice message. The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
  • Page 289 Configuring Advanced Features 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 entries, and then display the remote phone book entries on the phone user interface.
  • Page 290 Administrator’s Guide for SIP-T4X IP Phones Specify how often the IP phone refreshes the local cache of the remote phone book. Parameter: features.remote_phonebook.flash_tim Specify whether to refresh the local cache of the remote phone book at a time when accessing the remote phone book.
  • Page 291 Configuring Advanced Features Parameters Permitted Values Default Directory->Remote Phone Book->Remote URL Phone User Interface: None String within 99 remote_phonebook.data.X.name Blank characters (X ranges from 1 to 5) Description: Configures the display name of the remote phone book item. Example: remote_phonebook.data.1.name = Test Web User Interface: Directory->Remote Phone Book->Display Name Phone User Interface:...
  • Page 292 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default None 3600 0, Integer from 21600 features.remote_phonebook.flash_time to 2592000 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 293 Configuring Advanced Features Enter the name in the Display Name field. Click Confirm to accept the change. To configure the remote phone book 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.
  • Page 294 Administrator’s Guide for SIP-T4X IP Phones The biggest plus for LDAP is that users can access the central LDAP directory of the corporation using IP phones. Therefore they do not have to maintain the local directory. Users can search and dial out from the LDAP directory and save LDAP entries to the local directory.
  • Page 295 Configuring Advanced Features ldap.number_filter ldap.tls_mode ldap.host ldap.port ldap.base ldap.user ldap.password ldap.max_hits ldap.name_attr ldap.numb_attr ldap.display_name ldap.version ldap.call_in_lookup ldap.call_out_lookup ldap.ldap_sort Assign an LDAP key. Parameter: linekey.X.type/ programablekey.X.type Configure the LDAP feature. Navigate to: http://<phoneIPAddress>/servl et?p=contacts-LDAP&q=load Web User Interface Assign an LDAP key. Local Navigate to: http://<phoneIPAddress>/servl...
  • Page 296 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Enables or disables LDAP feature on the IP phone. 0-Disabled 1-Enabled Web User Interface: Directory->LDAP->Enable LDAP Phone User Interface: None ldap.name_filter String within 99 characters Blank Description: Configures the criteria for searching the LDAP contact name attributes. The “*”...
  • Page 297 Configuring Advanced Features Parameters Permitted Values Default None ldap.tls_mode Integer from 0 to 2 Description: Configures the connection mode between the LDAP server and the IP phone. 0-LDAP—Unencrypted connection between LDAP server and the IP phone. (port 389 is used by default). 1-LDAP TLS Start—TLS/SSL connection between LDAP server and the IP phone (port 389 is used by default).
  • Page 298 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default ldap.base String within 99 characters Blank Description: Configures the LDAP search base which corresponds to the location of the LDAP phone book from which the LDAP search request begins. The search base narrows the search scope and decreases directory search time.
  • Page 299 Configuring Advanced Features Parameters Permitted Values Default None ldap.max_hits Integer from 1 to 32000 Description: Configures the maximum number of search results to be returned by the LDAP server. If the value of the “Max.Hits” is blank, the LDAP server will return all searched results. Please note that a very large value of the “Max.
  • Page 300 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Configures the number attributes of each record to be returned by the LDAP server. You can configure multiple number attributes separated by spaces. Example: ldap.numb_attr = telephoneNumber Web User Interface: Directory->LDAP->LDAP Number Attributes...
  • Page 301 Configuring Advanced Features Parameters Permitted Values Default Description: Enables or disables the IP phone to perform an LDAP search when receiving an incoming call. 0-Disabled 1-Enabled Web User Interface: Directory->LDAP->LDAP Lookup For Incoming Call Phone User Interface: None ldap.call_out_lookup 0 or 1 Description: Enables or disables the IP phone to perform an LDAP search when placing a call.
  • Page 302 Administrator’s Guide for SIP-T4X IP Phones LDAP Key For more information on how to configure the DSS Key, refer to Appendix C: Configuring DSS Key on page 514. Permitted Parameter Default Values Refer to the linekey.X.type/programablekey.X.type following content Description: Configures a DSS key as an LDAP key on the IP phone.
  • Page 303 Configuring Advanced Features Permitted Parameter Default Values When X=2, the default value is 61 ( Directory When X=3, the default value is 5 ( 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 304 Administrator’s Guide for SIP-T4X IP Phones Select the desired values from the corresponding pull-down lists. Click Confirm to accept the change. To configure an LDAP key via web user interface: Click on DSSKey->Line Key X. In the desired DSS key field, select LDAP from the pull-down list of Type.
  • Page 305 Configuring Advanced Features (Optional.) Enter the string that will appear on the LCD screen in the Label field. Press the Save soft key to accept the change. Busy Lamp Field (BLF) is used to monitor a specific user for status changes on IP phones. For example, you can configure a BLF key on a supervisor’s phone to monitor the IP phone user status (busy or idle).
  • Page 306 Administrator’s Guide for SIP-T4X IP Phones LED Status Description The monitored user’s conversation is placed on hold (This LED status requires server support). The call is parked against the monitored user’s Slow flashing red (1s) phone number. The monitored user does not exist.
  • Page 307 Configuring Advanced Features LED Status Description The call is parked against the monitored user’s Slow flashing red (1s) phone number. The monitored user is idle. The monitored user does not exist. Note BLF LED Mode feature is only applicable to IP phones running firmware version 72 or later.
  • Page 308 Administrator’s Guide for SIP-T4X IP Phones Phone User Assign a BLF key. Interface Details of Configuration Parameters: Parameters Permitted Values Default features.pickup.blf_visual_enable 0 or 1 Description: Enables or disables the IP phone to display a visual alert when the monitored user receives an incoming call.
  • Page 309 Configuring Advanced Features Parameters Permitted Values Default None BLF Key For more information on how to configure the DSS Key, refer to Appendix C: Configuring DSS Key on page 514. Parameters Permitted Values Default Refer to the linekey.X.type following content Description: Configures a line key to be a BLF key on the IP phone.
  • Page 310 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default 1-6 correspond linekey.X.line Integer from 1 to 16 to the lines 1-6. Description: Configures the desired line to apply the BLF key. X ranges from 1 to 29 (for SIP-T48G).
  • Page 311 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the pickup code for BLF feature. This parameter only applies to BLF feature. X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). Example: linekey.3.pickup_value = *97 Web User Interface:...
  • Page 312 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Audio Alert for BLF Pickup. Click Confirm to accept the change. To configure BLF LED mode via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of BLF LED mode.
  • Page 313 Configuring Advanced Features (Optional.) Enter the string that will appear on the LCD screen in the Label field. Enter the phone number or extension you want to monitor in the Value field. (Optional.) Enter the directed call pickup code in the Extension field. Press the Save soft key to accept the change.
  • Page 314 Administrator’s Guide for SIP-T4X IP Phones Configure BLF List. http://<phoneIPAddress>/servlet?p=ac count-adv&q=load&acc=0 Web User Interface Assign a BLF List key. Navigate to: Local http://<phoneIPAddress>/servlet?p=ds skey&model=1&q=load&linepage=1 Phone User Assign a BLF List key. Interface Details of Configuration Parameters: Parameters Permitted Values Default String within 256 account.X.blf.blf_list_uri...
  • Page 315 Configuring Advanced Features Parameters Permitted Values Default Web User Interface: Account->Advanced->BLF List Code Phone User Interface: None String within 32 account.X.blf_list_barge_in_code Blank characters Description: Configures the barge-in code for account X. X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P).
  • Page 316 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Enables or disables the IP phone to automatically configure the BLF list keys. 0-Disabled 1-Enabled Web User Interface: None Phone User Interface: None phone_setting.blf_list_sequence_type 0 or 1 Description: Configures the order of BLF list keys assigned automatically.
  • Page 317 Configuring Advanced Features Parameters Permitted Values Default Example: linekey.2.type = 39 Default: For SIP-T48G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0. For SIP-T46G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
  • Page 318 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Line Key X/Programable Key DSSKey-> ->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. Select the desired account from the pull-down list of Account.
  • Page 319 Configuring Advanced Features Select the desired line from the pull-down list of Line. Repeat step 2-3, configure more BLF list keys. Click Confirm to accept the change. Hide Features Access Code feature enables the IP phone to display the feature identifier instead of the dialed feature access code automatically.
  • Page 320 Administrator’s Guide for SIP-T4X 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 321 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 322 Administrator’s Guide for SIP-T4X 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...
  • Page 323 Configuring Advanced Features Parameters Permitted Values Default Web User Interface: Features->ACD->ACD Auto Available Phone User Interface: None acd.auto_available_timer Integer from 0 to 120 Description: Configures the length of time (in seconds) before the status of the ACD agent is automatically changed to available. Web User Interface: Features->ACD->ACD Auto Available Timer (0~120s) Phone User Interface:...
  • Page 324 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P). Web User Interface: None Phone User Interface: None ACD Key For more information on how to configure the DSS Key, refer to...
  • Page 325 Configuring Advanced Features Parameter Permitted Values Default Web User Interface: DSSKey->Line Key X->Line KeyX->Type Phone User Interface: Menu->Features->DSS Keys->Line Key X->Type To configure an ACD key via web user interface: Click on DSSKey->Line Key X. In the desired DSS key field, select ACD from the pull-down list of Type. Click Confirm to accept the change.
  • Page 326 Administrator’s Guide for SIP-T4X 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.
  • Page 327 Configuring Advanced Features Procedure Configuration changes can be performed using the configuration files or locally. Configure subscribe for MWI. Parameters: account.X.subscribe_mwi account.X.subscribe_mwi_expires account.X.subscribe_mwi_to_vm Configure subscribe MWI to voice Configuration File <MAC>.cfg mail. Parameter: voice_mail.number.X Configure the presentation of audio and visual MWI. Parameter: account.X.display_mwi.enable Configure subscribe for MWI.
  • Page 328 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P). Web User Interface: Account->Advanced->Subscribe for MWI Phone User Interface: None account.X.subscribe_mwi_expires Integer from 0 to 84600...
  • Page 329 Configuring Advanced Features Parameters Permitted Values Default Account->Advanced->Subscribe MWI To Voice Mail Phone User Interface: None String within 99 voice_mail.number.X Blank characters Description: Configures the voice mail number for account X. X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P).
  • Page 330 Administrator’s Guide for SIP-T4X IP Phones To configure subscribe for MWI via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced. Select the desired value from the pull-down list of Subscribe for MWI.
  • Page 331 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 332 Administrator’s Guide for SIP-T4X IP Phones Multicast paging allows IP phones to send/receive Real-time Transport Protocol (RTP) streams to/from the pre-configured multicast address(es) without involving SIP signaling. Up to 10 listening multicast addresses can be specified on the IP phone.
  • Page 333 Configuring Advanced Features multicast.paging_address.X.label Assign a multicast paging key or a paging list key. Navigate to: http://<phoneIPAddress>/servlet ?p=dsskey&model=1&q=load&li nepage=1 Specify a multicast codec for the IP phone to send the RTP stream. Navigate to: Web User Interface http://<phoneIPAddress>/servlet ?p=features-general&q=load Local Configure the multicast IP address and port number for a paging list key.
  • Page 334 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default multicast.paging_address.X.ip_address String Blank Description: Configures the multicast IP address and port number for a paging list key. X ranges from 1 to 10. Example: multicast.paging_address.1.ip_address = 224.5.6.20:10008 Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
  • Page 335 Configuring Advanced Features Parameters Permitted Values Default X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). Example: linekey.2.type = 24 Default: For SIP-T48G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0.
  • Page 336 Administrator’s Guide for SIP-T4X IP Phones Paging List key For more information on how to configure the DSS Key, refer to Appendix C: Configuring DSS Key on page 514. Permitted Parameters Default Values Refer to the following linekey.X.type/programablekey.X.type content Description: Configures a DSS key as a paging list key on the IP phone.
  • Page 337 Configuring Advanced Features Permitted Parameters Default Values When X=3, the default value is 5 ( 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...
  • Page 338 Administrator’s Guide for SIP-T4X IP Phones The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255. Click Confirm to accept the change. To configure a paging list key via web user interface: Click on DSSKey->Line Key X. In the desired DSS key field, select Paging List from the pull-down list of Type.
  • Page 339 Configuring Advanced Features Select the desired codec from the pull-down list of Multicast Codec. Click Confirm to accept the change. To configure multicast addresses for a paging list key via web user interface: Click on Directory->Multicast IP. Enter the multicast address and port number in the Paging Address field. Enter the label in the Label field.
  • Page 340 Administrator’s Guide for SIP-T4X IP Phones 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.
  • Page 341 Configuring Advanced Features will be automatically ignored. If the parameter is the priority value, the incoming multicast paging calls with higher priority are automatically answered and the ones with lower priority are ignored. Paging Priority Active This parameter decides how the IP phone handles the incoming multicast paging calls when there is already a multicast paging call in progress.
  • Page 342 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Example: multicast.listen_address.1.ip_address = 224.5.6.20:10008 Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255. Web User Interface: Directory->Multicast IP->Multicast Listening->Listening Address Phone User Interface: None multicast.listen_address.X.label String within 99 characters...
  • Page 343 Configuring Advanced Features Parameters Permitted Values Default Configures the priority of multicast paging calls. 1 is the highest priority, 10 is the lowest priority. If it is set to 0, all incoming multicast paging calls will be automatically ignored. Web User Interface: Directory->Multicast IP->Multicast Listening->Paging Barge Phone User Interface: None...
  • Page 344 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Paging Priority Active. Click Confirm to accept the change. Call recording enables users to record calls. It depends on support from a SIP server. When the user presses the call record key, the IP phone sends a record request to the server.
  • Page 345 Get /phonerecording.cgi?model=yealink HTTP/1.0\r\n Request Method: GET Request URI: /phonerecording.cgi?model=yealink Request version: HTTP/1.0 Host: 10.1.2.224\r\n User-agent: yealink SIP-T46G 28.72.0.1 00:16:65:11:30:68\r\n If the recording is successfully started, the server will respond with a 200 OK message. Example of a 200 OK message: <YealinkIPPhoneText> <Title>...
  • Page 346 Administrator’s Guide for SIP-T4X IP Phones <YealinkIPPhoneText> <Title> </Title> <Text> Probably the recording box is full. </Text> <YealinkIPPhoneText> When the user presses the URL record key for the second time, the IP phone sends an HTTP GET message to the server, and then the server will respond with a 200 OK message.
  • Page 347 Configuring Advanced Features Record Key For more information on how to configure the DSS Key, refer to Appendix C: Configuring DSS Key on page 514. Parameters Permitted Values Default Refer to the linekey.X.type following content Description: Configures a line key to be a record key on the IP phone. The digit 25 stands for the key type Record.
  • Page 348 Administrator’s Guide for SIP-T4X IP Phones URL Record Key For more information on how to configure the DSS Key, refer to Appendix C: Configuring DSS Key on page 514. Parameters Permitted Values Default Refer to the linekey.X.type following content Description: Configures a line key to be a URL record key on the IP phone.
  • Page 349 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the URL to record a call. X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). Example: linekey.2.value = http://10.1.2.224/phonerecording.cgi Web User Interface: DSSKey->Line Key X->Line KeyX->Value...
  • Page 350 Administrator’s Guide for SIP-T4X IP Phones Enter the URL in the Value 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 351 Configuring Advanced Features hot desking key. Procedure Hot desking key can be configured using the configuration files or locally. Assign a hot desking key. Parameter: Configuration File <y0000000000xx>.cfg linekey.X.type/ programablekey.X.type Assign a hot desking key. Navigate to: Web User Interface http://<phoneIPAddress>/servlet Local ?p=dsskey&q=load&model=1...
  • Page 352 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameters Default Values For SIP-T48G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-29 is 0. For SIP-T46G IP phones: The default value of the line key 1-16 is 15, and the default value of the line key 17-27 is 0.
  • Page 353 Configuring Advanced Features Permitted Parameters Default Values When X=8, the default value is 52 ( Switch Account Down When X=9, the default value is 33 (Status). When X=10, the default value is 0 ( When X=13, the default value is 0 ( Web User Interface: /Programable Key DSSKey->Line Key X...
  • Page 354 Administrator’s Guide for SIP-T4X IP Phones HTTP or HTTPS GET request. You can specify a URL that triggers a GET request when a specified event occurs. Action URL can only be triggered by the pre-defined events (e.g., log on). The valid URL format is: http(s)://IP address of the server/help.xml?.
  • Page 355 Configuring Advanced Features Event Description Forward Incoming Call When the IP phone forwards an incoming call. Reject Incoming Call When the IP phone rejects an incoming call. Answer New-In Call When the IP phone answers a new call. Transfer Finished When the IP phone completes to transfer a call.
  • Page 356 Administrator’s Guide for SIP-T4X IP Phones Variable Value Description an incoming call. The SIP URI of the callee when the IP phone places a call. $remote The SIP URI of the caller when the IP phone receives an incoming call.
  • Page 357 Configuring Advanced Features action_url.transfer_call action_url.blind_transfer_call action_url.attended_transfer_call action_url.hold action_url.unhold action_url.mute action_url.unmute action_url.missed_call action_url.call_terminated action_url.busy_to_idle action_url.idle_to_busy action_url.ip_change action_url.forward_incoming_call action_url.reject_incoming_call action_url.answer_new_incoming_call action_url.transfer_finished action_url.transfer_failed action_url.setup_autop_finish Configure the action URL. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet?p=f eatures-actionurl&q=load Details of Configuration Parameters: Parameters Permitted Values Default action_url.setup_completed URL within 511 characters...
  • Page 358 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default $firmware  $active_url  $active_user  $active_host  $local  $remote  $display_local  $display_remote  $call_id  Example: action_url. setup_completed = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Setup Completed action_url.registered...
  • Page 359 Configuring Advanced Features Parameters Permitted Values Default Features->Action URL->Unregistered Phone User Interface: None action_url.register_failed URL within 511 characters Blank Description: Configures the action URL the IP phone sends after a register failed. Example: action_url.register_failed = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Register Failed Phone User Interface: None action_url.off_hook...
  • Page 360 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Phone User Interface: None action_url.incoming_call Blank URL within 511 characters Description: Configures the action URL the IP phone sends when receiving an incoming call. Example: action_url.incoming_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Incoming Call...
  • Page 361 Configuring Advanced Features Parameters Permitted Values Default None action_url.dnd_on URL within 511 characters Blank Description: Configures the action URL the IP phone sends when DND feature is enabled. Example: action_url.dnd_on = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Open DND Phone User Interface: None action_url.dnd_off URL within 511 characters...
  • Page 362 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default None action_url.always_fwd_off Blank URL within 511 characters Description: Configures the action URL the IP phone sends when always forward feature is disabled. Example: action_url.always_fwd_off = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Close Always Forward...
  • Page 363 Configuring Advanced Features Parameters Permitted Values Default Phone User Interface: None action_url.no_answer_fwd_on Blank URL within 511 characters Description: Configures the action URL the IP phone sends when no answer forward feature is enabled. Example: action_url.no_answer_fwd_on = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Open No Answer Forward Phone User Interface: None...
  • Page 364 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Phone User Interface: None action_url.blind_transfer_call Blank URL within 511 characters Description: Configures the action URL the IP phone sends when performing a blind transfer. Example: action_url.blind_transfer_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Blind Transfer...
  • Page 365 Configuring Advanced Features Parameters Permitted Values Default None action_url.unhold Blank URL within 511 characters Description: Configures the action URL the IP phone sends when resuming a held call. Example: action_url.unhold = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->UnHold Phone User Interface: None action_url.mute Blank...
  • Page 366 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default action_url.missed_call Blank URL within 511 characters Description: Configures the action URL the IP phone sends when missing a call. Example: action_url.missed_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Missed Call Phone User Interface: None action_url.call_terminated...
  • Page 367 Configuring Advanced Features Parameters Permitted Values Default action_url.idle_to_busy Blank URL within 511 characters Description: Configures the action URL the IP phone sends when changing the state of the IP phone from idle to busy. Example: action_url.idle_to_busy = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Idle To Busy Phone User Interface: None...
  • Page 368 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default action_url.reject_incoming_call Blank URL within 511 characters Description: Configures the action URL the IP phone sends when rejecting an incoming call. Example: action_url.reject_incoming_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Reject Incoming Call...
  • Page 369 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the action URL the IP phone sends when failing to transfer a call. Example: action_url.transfer_failed = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Transfer Failed Phone User Interface: None action_url.setup_autop_finish URL within 511 characters Blank Description: Configures the action URL the IP phone sends when completing auto provisioning...
  • Page 370 Administrator’s Guide for SIP-T4X IP Phones Enter the action URLs in the corresponding fields. Click Confirm to accept the change. Opposite to action URL, action URI allows IP phones to interact with web server application by receiving and handling an HTTP or HTTPS GET request. When receiving a GET request, the IP phone will perform the specified action and respond with a 200 OK message.
  • Page 371 Configuring Advanced Features Variable Value Phone Action Cancel actions or reject incoming calls. CANCEL Return to a previous screen or cancel a call. 0-9/*/POUND Press the keypad (0-9, * or #). Press the line key (for SIP-T48G, X=29, L1-LX SIP-T46G, X=27, for SIP-T42G/T41P , X=15). F_CONFERENCE Press the Conference soft key.
  • Page 372 Administrator’s Guide for SIP-T4X IP Phones Variable Value Phone Action Get firmware version, registration, DND or forward configuration information. The valid value of “x” is 0 or 1, 0 means you do not need to get configuration information. 1 means you want to get configuration information.
  • Page 373 Configuring Advanced Features et?p=features-remotecontrl&q =load Details of the Configuration Parameter: Parameter Permitted Values Default features.action_uri_limit_ip IP address or any Blank Description: Configures the address(es) from which Action URI will be accepted. For discontinuous IP addresses, multiple IP addresses are separated by commas. For continuous IP addresses, the format likes *.*.*.* and the “*”...
  • Page 374 Administrator’s Guide for SIP-T4X IP Phones 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.
  • Page 375 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...
  • Page 376 Working Server: Server 1 is configured with the domain name of the working server. For example, yealink.pbx.com. DNS mechanism is used such that the working server is resolved to multiple physical SIP servers for failover purpose. The working server is deployed in redundant pairs, designated as primary and secondary servers.
  • Page 377 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 378 Administrator’s Guide for SIP-T4X IP Phones Details of Configuration Parameters: Parameters Permitted Values Default account.X.sip_server.Y.address String within 256 Blank characters (X ranges from 1 to 16. Y ranges from 1 to 2) Description: Configures the IP address or domain name of the SIP server Y for account X.
  • Page 379 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the registration expiration time (in seconds) of the SIP server Y for account X. X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P). Example: account.1.sip_server.1.expires = 3600 Web User Interface:...
  • Page 380 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Web User Interface: None Phone User Interface: None Integer from 10 to account.X.fallback.timeout 2147483647 Description: Configures the time interval (in seconds) for the IP phone to detect whether the working server is available by sending the registration request after the fallback server takes over call control.
  • Page 381 Configuring Advanced Features Parameters Permitted Values Default X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P). Web User Interface: None Phone User Interface: None account.X.sip_server.Y.failback_timeout 0, 60 to 65535 3600 (X ranges from 1 to 16, Y ranges from 1 to 2) Description: Configures the time (in seconds) for the phone to retry to send requests to the primary server after failing over to the current working server when the parameter...
  • Page 382 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Phone User Interface: None To configure server redundancy for fallback purpose via web user interface: Click on Account->Register. Select the desired account from the pull-down list of Account. Select the desired value from the pull-down list of Transport.
  • Page 383 Configuring Advanced Features You must set the port of SIP server to 0 for NAPTR, SRV and A queries. Click Confirm to accept the change. Note If the outbound proxy server is required and the transport is set to DNS-NAPTR, you must set the port of outbound proxy server to 0 for NAPTR, SRV and A queries.
  • Page 384 Administrator’s Guide for SIP-T4X IP Phones The following details the procedures of DNS query for the IP phone to resolve the domain name (e.g., yealink.pbx.com) of working server into the IP address, port and transport protocol. NAPTR (Naming Authority Pointer) First, the IP phone sends the NAPTR query to get the SRV pointer and and transport protocol.
  • Page 385 The two records also contain a port “5060”, the IP phone uses this port. If the Target is not a numeric IP address, the IP phone performs the A query. So in this case, the IP phone uses “server1.yealink.pbx.com” and “server2.yealink.pbx.com" for the A query.
  • Page 386 Administrator’s Guide for SIP-T4X IP Phones If it is not the last server in the list, the maximum number of retries depends on the configured retry count. Procedure SIP Server Domain Name Resolution can be configured using the configuration files or locally.
  • Page 387 Configuring Advanced Features Parameters Permitted Values Default account.X.naptr_build 0 or 1 Description: Configures the way of SRV query for the IP phone to be performed when no result is returned from NAPTR query for account X. 0-SRV query using UDP only 1-SRV query using UDP, TCP and TLS X ranges from 1 to 16 (for SIP-T48G/T46G).
  • Page 388 Administrator’s Guide for SIP-T4X IP Phones IP phones can be configured to use static DNS cache preferentially. Static DNS cache is configurable on a per-line basis. Procedure Static DNS cache can be configured only using the configuration files. Configure NAPTR/SRV/A records.
  • Page 389 String within 256 Blank characters (X ranges from 1 to 12) Description: Configures the domain name to which NAPTR record X refers. Example: dns_cache_naptr.1.name = yealink.pbx.com Web User Interface: None Phone User Interface: None dns_cache_naptr.X.flags S, A, U or P...
  • Page 390 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default dns_cache_naptr.1.order = 90 Web User Interface: None Phone User Interface: None dns_cache_naptr.X.preference Integer from 0 to 65535 (X ranges from 1 to 12) Description: Configures the preference of NAPTR record X. NAPTR record with lower preference is more preferred.
  • Page 391 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)
  • Page 392 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Configures the port to be used in SRV record X. Example; dns_cache_srv.1.port = 5060 Web User Interface: None Phone User Interface: None dns_cache_srv.X.priority Integer from 0 to 65535 (X ranges from 1 to 12) Description: Configures the priority for the target host in SRV record X.
  • Page 393 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 394 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Web User Interface: None Phone User Interface: None dns_cache_a.X.ttl Integer from 30 to 2147483647 (X ranges from 1 to 12) Description: Configures the time interval (in seconds) that A record X may be cached before the record should be consulted again.
  • Page 395 Configuring Advanced Features Parameters Permitted Values Default Description: Configures whether preferentially to use the static DNS cache for domain name resolution of the SIP server for account X. 0-Use domain name resolution from the DNS server preferentially 1-Use static DNS cache preferentially X ranges from 1 to 16 (for SIP-T48G/T46G).
  • Page 396 Administrator’s Guide for SIP-T4X IP Phones Inventory Management -- provides a means to effectively manage IP phones and  their attributes such as model number, serial number and software revision. TLVs supported by IP phones are summarized in the following table:...
  • Page 397 Inventory – Serial Serial number of phone. Number Inventory – Manufacturer name of phone. Manufacturer Name The default value is “yealink”. Inventory – Model Model name of phone. Name Assertion identifier of phone. Asset ID The default value is “asset”.
  • Page 398 Administrator’s Guide for SIP-T4X 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 399 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 the settings will take effect after reboot. Click OK to reboot the phone. To configure LLDP feature via phone user interface: Press Menu->Advanced (default password: admin) ->Network->LLDP->LLDP Status.
  • Page 400 Administrator’s Guide for SIP-T4X IP Phones tagged packets to the corresponding VLAN according to the VLAN ID in the tag as described in IEEE Std 802.3. VLAN on IP phones allows simultaneous access for a regular PC. This feature allows a PC to be daisy chained to an IP phone and the connection for both PC and IP phone to be trunked through the same physical Ethernet cable.
  • Page 401 Configuring Advanced Features port and PC port. Configure DHCP VLAN discovery feature. Navigate to: http://<phoneIPAddress>/servlet? p=network-adv&q=load Configure VLAN for the Internet port and PC port. Phone User Interface Configure DHCP VLAN discovery feature. Details of Configuration Parameters: Parameters Permitted Values Default network.vlan.internet_port_enable 0 or 1...
  • Page 402 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Port->VID Number 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 403 Configuring Advanced Features Parameters Permitted Values Default Phone User Interface: Menu->Advanced (default password: admin) ->Network->VLAN ->PC Port->VID network.vlan.pc_port_priority Integer from 0 to 7 Description: Configures VLAN priority for the PC (LAN) port. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 404 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default 0-Disabled 1-Enabled Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Advanced->VLAN >DHCP VLAN->Active Phone User Interface: Menu->Advanced (default password: admin) ->Network->VLAN ->DHCP VLAN ->DHCP VLAN...
  • Page 405 Configuring Advanced Features In the VLAN block, select the desired value from the pull-down list of WAN Port Active. Enter the VLAN ID in the VID (1-4094) field. 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 reboot to make the settings effective.
  • Page 406 Administrator’s Guide for SIP-T4X IP Phones In the VLAN block, select the desired value from the pull-down list of DHCP VLAN Active. Enter the desired option in the Option (1-255) field. The default option is 132. Click Confirm to accept the change.
  • Page 407 The file format of the compressed package must be *.tar. The VPN-related files are: certificates (ca.crt and client.crt), key (client.key) and the configuration file (vpn.cnf) of the VPN client. For more information on how to package a OpenVPN Feature on Yealink IP Phones TAR file, refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 408 Administrator’s Guide for SIP-T4X IP Phones Navigate to: http://<phoneIPAddress>/servl et?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: Enables or disables OpenVPN feature on the IP phone. 0-Disabled 1-Enabled Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 409 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 reboot. Click OK to reboot the phone.
  • Page 410 Administrator’s Guide for SIP-T4X IP Phones be considered when configuring a modern QoS implementation: bandwidth, delay, jitter and loss. QoS provides better network service through the following features: Supporting dedicated bandwidth  Improving loss characteristics  Avoiding and managing network congestion ...
  • Page 411 Configuring Advanced Features Voice QoS In order to make VoIP transmissions intelligible to receivers, voice packets should not be dropped, excessively delayed, made to suffer varying delay. DiffServ model can guarantee high-quality voice transmission when the voice packets are configured to a higher DSCP value.
  • Page 412 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Phone User Interface: None network.qos.signaltos Integer from 0 to 63 Description: Configures the DSCP for SIP packets. The default DSCP value for SIP packets is 26 (Assured Forwarding). Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 413 Configuring Advanced Features Network Address Translation (NAT) is essentially a translation table that maps public IP address and port combinations to private ones. This reduces the need for a large number of public IP addresses. The NAT feature ensures security since each outgoing or incoming request must first go through a translation process.
  • Page 414 Administrator’s Guide for SIP-T4X IP Phones Details of Configuration Parameters: Parameters Permitted Values Default account.X.nat.nat_traversal 0 or 1 Description: Enables or disables the NAT traversal for account X. 0-Disabled 1-Enabled X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G).
  • Page 415 Configuring Advanced Features Parameters Permitted Values Default Example: account.1.nat.stun_port = 3478 Web User Interface: Account->Register ->STUN Server ->Port Phone User Interface: None To configure the NAT traversal and STUN server via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Select STUN from the pull-down list of NAT.
  • Page 416 Administrator’s Guide for SIP-T4X IP Phones the authentication server for verification. If the authentication server determines the credentials are valid, the IP phone is allowed to access resources located on the protected side of the network. IP phones support protocols EAP-MD5, EAP-TLS, EAP-PEAP/MSCHAPv2, EAP-TTLS/EAP-MSCHAPv2, EAP-PEAP/GTC and EAP-TTLS/EAP-GTC for 802.1X...
  • Page 417 String within 32 characters Blank Description: Configures the user name for 802.1x authentication. Example: network.802_1x.identity = yealink Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Advanced->802.1x->Identity Phone User Interface: Menu->Advanced (default password: admin) ->Network->802.1x >802.1x Mode...
  • Page 418 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default (EAP-MD5)->MD5 assword network.802_1x.root_cert_url URL within 511 characters Blank Description: Configures the access URL of the CA certificate when the 802.1x authentication method is configured as EAP-TLS, EAP-PEAP/MSCHAPv2, EAP-TTLS/EAP-MSCHAPV2, EAP-PEAP/GTC or EAP-TTLS/EAP-GTC Example : network.802_1x.root_cert_url = http://192.168.1.10/ca.pem...
  • Page 419 Configuring Advanced Features a) If you select EAP-MD5: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. b) If you select EAP-TLS: 1) Enter the user name for authentication in the Identity field. 2) Leave the MD5 Password field blank.
  • Page 420 Administrator’s Guide for SIP-T4X 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.
  • Page 421 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 locate the desired certificate (*.pem,*.crt, *.cer or *.der) from your local system.
  • Page 422 Administrator’s Guide for SIP-T4X IP Phones 4) Click Upload to upload the certificate. e) If you select EAP-PEAP/GTC: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field.
  • Page 423 Configuring Advanced Features 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system. 4) Click Upload to upload the certificate. f) If you select EAP-TTLS/EAP-GTC: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field.
  • Page 424 Administrator’s Guide for SIP-T4X IP Phones 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. Click Confirm to accept the change.
  • Page 425 Configuring Advanced Features c) If you select EAP-PEAP/MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. d) If you select EAP-TTLS/EAP-MSCHAPV2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field.
  • Page 426 Administrator’s Guide for SIP-T4X IP Phones RPC Method Description This method is used to obtain the value of one or GetParameterValues more CPE parameters. This method is used to discover the parameters GetParameterNames accessible on a particular CPE. This method is used to read the attributes associated GetParameterAttributes with one or more CPE parameters.
  • Page 427 Configuring Advanced Features Procedure TR-069 can be configured using the configuration files or locally. Configure theTR-069 feature. Parameters: managementserver.enable managementserver.username managementserver.password Configurati managementserver.url <y0000000000xx>.cfg on File managementserver.connection_request_use rname managementserver.connection_request_pas sword managementserver.periodic_inform_enable managementserver.periodic_inform_interval Configure the TR-069 feature. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet?p=settings -preference&q=load Details of Configuration Parameters:...
  • Page 428 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameters Default Values Configuration Servers). This string is set to the empty string if no authentication is required. Example: managementserver.username = user1 Web User Interface: Settings->TR069->ACS Username Phone User Interface: None String within managementserver.password...
  • Page 429 Configuring Advanced Features Permitted Parameters Default Values Configures the user name for the IP phone to authenticate the incoming connection requests. Example: managementserver.connection_request_username = accuser Web User Interface: Settings->TR069->Connection Request Username Phone User Interface: None String within managementserver.connection_request_password Blank 64 characters Description: Configures the password for the IP phone to authenticate the incoming connection requests.
  • Page 430 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameters Default Values Configures the interval (in seconds) for the IP phone to report its configuration to the ACS (Auto Configuration Servers). Web User Interface: Settings->TR069->Periodic Inform Interval (seconds) Phone User Interface: None To configure TR-069 via web user interface: Click on Settings->TR069.
  • Page 431 Configuring Advanced Features address, consisting of eight groups of four hexadecimal digits separated by colons. VoIP network based on IPv6 can ensure QoS, a set of service requirements to deliver performance guarantee while transporting traffic over the network. IPv6 Address Assignment Method Supported IPv6 address assignment methods: Manual Assignment: An IPv6 address and other configuration parameters (e.g., ...
  • Page 432 Administrator’s Guide for SIP-T4X IP Phones network.ipv6_icmp_v6.enable Configure the IPv6 static DNS address. network.ipv6_primary_dns network.ipv6_secondary_dns Configure the IPv6 static DNS. <y0000000000xx>.cf Parameter: network.ipv6_static_dns_enable Configure the IPv6 address assignment method. Configure the IPv6 static DNS. Web User Interface Navigate to: http://<phoneIPAddress>/servlet?p Local =network&q=load...
  • Page 433 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the Internet (WAN) port type for IPv6 when the IP address mode is configured as IPv6 or IPv4&IPv6. 0-DHCP 1-Static IP Address Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 434 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Web User Interface: Network->Basic->IPv6 Config->Static IP Address->IP Address Phone User Interface: Menu->Advanced (default password: admin) ->Network->WAN Port->IPv6->Type(Static IP) ->IP Address network.ipv6_prefix Integer from 0 to 128 Description: Configures the IPv6 prefix when the IP address mode is configured as IPv6 or IPv4&IPv6, and the Internet (WAN) port type for IPv6 is configured as Static IP...
  • Page 435 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the primary IPv6 DNS server when the IP address mode is configured as IPv6 or IPv4&IPv6, and the Internet (WAN) port type for IPv6 is configured as Static IP Address, or and the Internet (WAN) port type for IPv6 is configured as DHCP and Staic DNS is configured as Enabled.
  • Page 436 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Description: Enables or disables the IP phone to obtain IPv6 network settings via SLAAC (Stateless Address Autoconfiguration) method. 0-Disabled 1-Enabled Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 437 Configuring Advanced Features (Optional.) If you mark the DHCP radio box, you can configure the static DNS address in the corresponding fields. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the phone.
  • Page 438 Administrator’s Guide for SIP-T4X 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 via phone user interface: Press Menu->Advanced (default password: admin) ->Network->WAN Port.
  • Page 439 Configuring Audio Features This chapter provides information for making configuration changes for the following audio features: Headset Prior  Dual Headset  Voice Quality Monitoring  Audio Codecs  Acoustic Clarity Technology  Headset prior allows users to use headset preferentially if a headset is physically connected to the IP phone.
  • Page 440 Administrator’s Guide for SIP-T4X IP Phones Parameter Permitted Values Default If it is set to 1 (enabled), a user needs to press the HEADSET key to activate the headset mode. The headset mode will not be deactivated until the user presses the HEADSET key again.
  • Page 441 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>/servlet ?p=features-general&q=load Details of the Configuration Parameter: Parameter Permitted Values Default features.headset_training...
  • Page 442 These metrics can be sent between the phones in RTCP-XR packets. These metrics can also be sent in SIP PUBLISH messages to a central voice quality report collector. Two methods for voice quality monitoring are supported by Yealink IP phones: RTCP-XR ...
  • Page 443 Configuring Audio Features Procedure RTCP-XR can be configured using the configuration files. Configure the generation of RTCP-XR packets. Configuration <y0000000000xx>.cfg File Parameter: phone_setting.rtcp_xr_report.enable Details of Configuration Parameters: 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 party during a call for call quality monitoring and diagnosing.
  • Page 444 Administrator’s Guide for SIP-T4X IP Phones Users can check the voice quality data of the last call via web user interface or phone user options of the RTP status to be displayed on the interface. Users can also specify the phone user interface.
  • Page 445 Configuring Audio Features showing the voice quality report of the last call or the current call on the phone user interface. Parameter: phone_setting.vq_rtcpxr.states_show_on_gui.e nable Configure the options of the RTP status displayed on the phone user interface. Parameters: phone_setting.vq_rtcpxr_display_start_time.en able phone_setting.vq_rtcpxr_display_stop_time.en able phone_setting.vq_rtcpxr_display_local_call_id.
  • Page 446 Administrator’s Guide for SIP-T4X IP Phones Interface Configure the phone to display RTP status showing the voice quality report of the last call on the web user interface. Configure the phone to display RTP status showing the voice quality report of the last call or the current call on the phone user interface.
  • Page 447 Configuring Audio Features Permitted Parameters Default Values Description: Enables or disables the IP phone to send an interval quality report to the central report collector periodically throughout a call. 0-Disabled 1-Enabled Web User Interface: Settings->Voice Monitoring->VQ RTCP-XR Interval Report Phone User Interface: None Integer from phone_setting.vq_rtcpxr_interval_period...
  • Page 448 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameters Default Values 15 to 40 Blank phone_setting.vq_rtcpxr_moslq_threshold_critical Description: Configures the desired threshold value of listening MOS score (MOS-LQ) multiplied by 10. The threshold value of MOS-LQ causes the phone to send a critical alert quality report to the central report collector.
  • Page 449 Configuring Audio Features Permitted Parameters Default Values Description: Configures the threshold value of one way delay (in ms) that causes phone to send a critical alert quality report to the central report collector. For example, If it is set to 500, when the value of one way delay computed by the phone is greater than or equal to 500, the phone will send a critical alert quality report to the central report collector;...
  • Page 450 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameters Default Values Phone User Interface: None phone_setting.vq_rtcpxr_display_start_time.enable 0 or 1 Description: Enables or disables the phone to display Start Time on the LCD screen. 0-Disabled 1-Enabled Note: It works only if the value of the parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable”...
  • Page 451 Configuring Audio Features Permitted Parameters Default Values Web User Interface: Settings->Voice Monitoring->Report options on phone UI->Local User Phone User Interface: None phone_setting.vq_rtcpxr_display_remote_call_id.enable 0 or 1 Description: Enables or disables the phone to display Remote User on the LCD screen. 0-Disabled 1-Enabled Note: It works only if the value of the parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable”...
  • Page 452 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameters Default Values Description: Enables or disables the phone to display Remote Codec on the LCD screen. 0-Disabled 1-Enabled Note: It works only if the value of the parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”.

  • Page 453: Phone_Setting.vq_Rtcpxr_Display_Round_Trip_Delay.enable

    Configuring Audio Features Permitted Parameters Default Values phone_setting.vq_rtcpxr_display_packets_lost.enable 0 or 1 Description: Enables or disables the phone to display Packet lost on the LCD screen. 0-Disabled 1-Enabled Note: It works only if the value of the parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1”. Web User Interface: Settings->Voice Monitoring->Report options on phone UI->Packet lost Phone User Interface:...

  • Page 454: Phone_Setting.vq_Rtcpxr_Display_Moslq.enabl

    Administrator’s Guide for SIP-T4X IP Phones Permitted Parameters Default Values Phone User Interface: None phone_setting.vq_rtcpxr_display_moslq.enable 0 or 1 Description: Enables or disables the phone to display MOS-LQ on the LCD screen. 0-Disabled 1-Enabled Note: It works only if the value of the parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable”...

  • Page 455: Account.x.vq_Rtcpxr.collector_Server_Host

    Configuring Audio Features Permitted Parameters Default Values X ranges from 1 to 6 (for SIP-T41P). Web User Interface: Account->Advanced->VQ RTCP-XR Collector name Phone User Interface: None IPv4 account.X.vq_rtcpxr.collector_server_host Blank Address Description: Configures the IP address of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages for account X.
  • Page 456 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of VQ RTCP-XR Session Report. Click Confirm to accept the change. To configure interval report for VQ RTCP-XR via web user interface: Click on Settings->Voice Monitoring.
  • Page 457 Configuring Audio Features To configure alert report for VQ RTCP-XR via web user interface: Click on Settings->Voice Monitoring. Enter the desired value in the Warning threshold for Moslq field. Enter the desired value in the Critical threshold for Moslq field. Enter the desired value in the Warning threshold for Delay field.
  • Page 458 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Display Report options on Web. Click Confirm to accept the change. The RTP status will appear on the web user interface at the path: Status.
  • Page 459 Configuring Audio Features Select the desired value from the pull-down list of Display Report options on phone. Click Confirm to accept the change. The RTP status will appear on the phone user interface at the path: Menu->Status->More…. To configure the options of the RTP status displayed on the LCD screen via web user interface: Click on Settings->Voice Monitoring.
  • Page 460 Administrator’s Guide for SIP-T4X IP Phones 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 461 This can effectively reduce the frame size and the bandwidth required for audio transmission. The following table lists the audio codecs supported by SIP-T4X IP phones: Supported Audio Codecs Default Audio Codecs...
  • Page 462 Administrator’s Guide for SIP-T4X IP Phones Codec Algorithm Reference Bit Rate Sample Packetization Rate Time 20ms G726-24 G.726 RFC 3551 24 Kbps 8 Ksps 20ms G726-32 G.726 RFC 3551 32 Kbps 8 Ksps 20ms G726-40 G.726 RFC 3551 40 Kbps...
  • Page 463 Configuring Audio Features Codec Configuration Methods Priority RTPmap Configuration Files G726-24 Web User Interface Configuration Files G726-32 Web User Interface Configuration Files G726-40 Web User Interface Configuration Files iLBC Web User Interface Procedure Configuration changes can be performed using the configuration files or locally. Configure the codecs to use on a per-line basis.
  • Page 464 Administrator’s Guide for SIP-T4X IP Phones Details of the Configuration Parameter: Permitted Parameters Default Values account.X.codec.Y.enable Refer to the 0 or 1 following content (X ranges from 1 to 16. Y ranges from 1 to 11) Description: Enables or disables the specified codec for account X.
  • Page 465 Configuring Audio Features Permitted Parameters Default Values Permitted Values: PCMU, PCMA, G729, G722, G723_53, G723_63, G726-16, G726-24, G726-32, G726-40, iLBC Default: When Y=1, the default value is PCMU; When Y=2, the default value is PCMA; When Y=3, the default value is G723_53; When Y=4, the default value is G723_63;...
  • Page 466 Administrator’s Guide for SIP-T4X IP Phones Permitted Parameters Default Values When Y=5, the default value is 4; When Y=6, the default value is 1; When Y=7, the default value is 0; When Y=8, the default value is 0; When Y=9, the default value is 0;...
  • Page 467 Configuring Audio Features Permitted Parameters Default Values 0 (Disabled), account.X.ptime 10, 20, 30, 40, 50 or 60 Description: Configures the ptime (in milliseconds) for the codec for account X. X ranges from 1 to 16 (for SIP-T48G/T46G). X ranges from 1 to 12 (for SIP-T42G). X ranges from 1 to 6 (for SIP-T41P).
  • Page 468 Administrator’s Guide for SIP-T4X IP Phones Click Confirm to accept the change. To configure the Ptime on a per-line basis via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced.
  • Page 469 Configuring Audio Features Procedure AEC can be configured using the configuration files or locally. Configure AEC. Configuration File <y0000000000xx>.cfg Parameter: voice.echo_cancellation Configure AEC. Navigate to: Local Web User Interface http://<phoneIPAddress>/servl et?p=settings-voice&q=load Details of the Configuration Parameter: Parameter Permitted Values Default voice.echo_cancellation 0 or 1 Description:...
  • Page 470 Administrator’s Guide for SIP-T4X IP Phones Click Confirm to accept the change. Background noise suppression (BNS) is designed primarily for hands-free operation and reduces background noise to enhance communication in noisy environments. Automatic Gain Control (AGC) is applicable to hands-free operation and is used to keep audio output at nearly a constant level by adjusting the gain of signals in certain circumstances.
  • Page 471 Configuring Audio Features Details of the Configuration Parameter: Parameter Permitted Values Default voice.vad 0 or 1 Description: Enables or disables VAD (Voice Activity Detection) feature on the IP phone. 0-Disabled 1-Enabled Web User Interface: Settings->Voice->Echo Cancellation ->VAD Phone User Interface: None To configure VAD via web user interface: Click on Settings->Voice.
  • Page 472 Administrator’s Guide for SIP-T4X IP Phones an acceptable perceived QoS while simultaneously keeping transmission costs and bandwidth usage as low as possible. Procedure CNG can be configured using the configuration files or locally. Configure CNG. Configuration File <y0000000000xx>.cfg Parameter: voice.cng Configure CNG.
  • Page 473 Configuring Audio Features 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 474 Administrator’s Guide for SIP-T4X IP Phones Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load Details of Configuration Parameters: Parameters Permitted Values Default voice.jib.adaptive 0 or 1 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...
  • Page 475 Configuring Audio Features 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 value of the parameter “voice.jib.adaptive” is set to 0 (Fixed). Web User Interface: Settings->Voice->JITTER BUFFER->Normal Phone User Interface:...
  • Page 476 Administrator’s Guide for SIP-T4X IP Phones...
  • Page 477 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 478 Administrator’s Guide for SIP-T4X IP Phones SIP-T4X IP phones support TLS 1.0. A cipher suite is a named combination of authentication, encryption, and message authentication code (MAC) algorithms used to negotiate the security settings for a network connection using the TLS/SSL network protocol.
  • Page 479 *.pem and *.cer. A unique server certificate: It is installed by default and is unique to an IP phone (based on the MAC address) and issued by the Yealink Certificate Authority (CA). A generic server certificate: It is installed by default and is issued by the...
  • Page 480 Administrator’s Guide for SIP-T4X IP Phones Yealink Certificate Authority (CA). Only if no unique certificate exists, the IP phone may send a generic certificate for authentication. The IP phone can authenticate the server certificate based on the trusted certificates list.
  • Page 481 Configuring Security Features certificates. Parameter: phone_setting.reserve_certs_en able Configure TLS on a per-line basis. Navigate to: http://<phoneIPAddress>/servl et?p=account-register&q=load &acc=0 Configure the trusted certificates feature. Upload the trusted certificates. Local Web User Interface Navigate to: http://<phoneIPAddress>/servl et?p=trusted-cert&q=load Configure the server certificates feature. Upload the server certificates.
  • Page 482 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Account->Register ->Transport Phone User Interface: None security.trust_certificates 0 or 1 Description: Enables or disables the IP phone to only trust the server certificates in the Trusted Certificates list. 0-Disabled 1-Enabled...
  • Page 483 Configuring Security Features Parameters Permitted Values Default Description: Enables or disables the IP phone to mandatorily validate the CommonName or SubjectAltName of the certificate sent by the server. 0-Disabled 1-Enabled Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 484 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Security->Trusted Certificates->Load trusted certificates file Phone User Interface: None URL within 511 server_certificates.url Blank characters Description: Configures the access URL of the certificate the IP phone sends for authentication. Example: server_certificates.url = http://192.168.1.20/ca.pem...
  • Page 485 Configuring Security Features Click OK to reboot the phone. To configure TLS on a per-line basis via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Select TLS from the pull-down list of the Transport. Click Confirm to accept the change.
  • Page 486 Administrator’s Guide for SIP-T4X IP Phones Click Browse to locate the certificate (*.pem,*.crt, *.cer or *.der) from your local system. Click Upload to upload the certificate. To configure the server certificates feature via web user interface: Click on Security->Server Certificates.
  • Page 487 Configuring Security Features Click Browse to locate the certificate (*.pem or *.cer) from your local system. Click Upload to upload the certificate. The dialog box pops up to prompt “Success: The Server Certificate has been loaded! Rebooting, please wait…”. Secure Real-Time Transport Protocol (SRTP) encrypts RTP streams during VoIP phone calls to avoid interception and eavesdropping.
  • Page 488 Administrator’s Guide for SIP-T4X IP Phones The callee receives the INVITE message with the RTP encryption algorithm, and then answers the call by responding with a 200 OK message which carries the negotiated RTP encryption algorithm. Example of the RTP encryption algorithm carried in the SDP of the 200 OK message:...
  • Page 489 Configuring Security Features Details of the Configuration Parameter: Parameters Permitted Values Default account.X.srtp_encryption 0, 1 or 2 Description: Configures whether to use voice encryption service for account X. 0-Disabled 1-Optional 2-Compulsory If it is set to 1 (Optional), the IP phone will negotiate with the other IP phone what type of encryption to utilize for the session.
  • Page 490 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 491: Procedure To Encrypt Configuration Files

    Configuring Security Features For a Microsoft Windows platform, you can use a Yealink-supplied encryption tool "Config_Encrypt_Tool.exe" to encrypt the <y0000000000xx>.cfg and <MAC>.cfg files respectively. Note Yealink also supplies a configuration encryption tool (yealinkencrypt) for Linux platform if Yealink Configuration Encryption Tool User Guide required.
  • Page 492 Administrator’s Guide for SIP-T4X IP Phones 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 493 Configuring Security Features Procedure Encryption method and AES keys can be configured using the configuration files or locally. Configure the decryption method and AES keys. Parameters： auto_provision.aes_key_in_file Configuration File <y0000000000xx>.cfg auto_provision.aes_key_16.com auto_provision.aes_key_16.mac auto_provision.update_file_mode Configure the AES keys. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet? p=settings-autop&q=load...
  • Page 494 Administrator’s Guide for SIP-T4X IP Phones 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 495 Configuring Security Features Parameters Permitted Values Default 1-Enabled Web User Interface: None Phone User Interface: None To configure the 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. and the AES keys must be 16 characters and the supported characters contain: 0-9, A-Z, a-z following special characters are also supported: # $ % * +, - .
  • Page 496 Administrator’s Guide for SIP-T4X IP Phones...
  • Page 497 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 498 Administrator’s Guide for SIP-T4X IP Phones Softkey Layout Template  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 499 Resource Files The following shows an example of a replace rule file: <DialRule> <Data Prefix="1" Replace="05928665234" LineID=""/> <Data Prefix="2(xx)" Replace="002$1" LineID="0"/> <Data Prefix="5([6-9])(.)" Replace="3$2" LineID="1,2,3"/> <Data Prefix="0(.)" Replace="9$1" LineID="2"/> <Data Prefix="1009" Replace="05921009" LineID="1"/> </DialRule> The dial-now template helps with the creation of multiple dial-now rules. After setup, place the dial-now template to the provisioning server and specify the access URL in the configuration files.
  • Page 500 Administrator’s Guide for SIP-T4X IP Phones The following shows an example of a dial-now template: <DialNow> <Data DialNowRule="1234" LineID="1"/> <Data DialNowRule="52[0-6]" LineID="1"/> <Data DialNowRule="xxxxxx" LineID=""/> </DialNow> The softkey layout template allows assigning different soft key layouts to different call states. The call states include CallFailed, CallIn, Connecting, Dialing (not applicable to SIP-T48G), RingBack and Talking.
  • Page 501 Resource Files same string introduced above. Specify the values within double quotes. Save the change and place this file to the provisioning server. The following shows an example of the CallFailed template file: <CallFailed> <Disable> <Key Type="Empty"/> <Key Type="Switch"/> <Key Type="Cancel"/> </Disable>...
  • Page 502 Administrator’s Guide for SIP-T4X IP Phones and 4. 1 is the highest priority, 4 is the lowest. When enabling or disabling the desired directory list for Directory, the valid values  are 0 and 1. 0 stands for Disabled, 1 stands for Enabled.
  • Page 503 Resource Files Remote Phone Book and LDAP . You can configure the search source list in dialing using the supplied super search template (super_search.xml). After setup, place the super search template to the provisioning server and specify the access URL in the configuration files.
  • Page 504 Administrator’s Guide for SIP-T4X IP Phones <item id_name="remote_directory_search" display_name="Remote Phone Book" priority="3" enable="0" /> <item id_name="ldap_search" display_name="LDAP" priority="4" enable="0" /> </root_super_search> You can add contacts one by one on the IP phone directly. You can also add multiple contacts at a time and/or share contacts between IP phones using the local contact template file.
  • Page 505 Resource Files 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=”” ring=””/> Where: display_name=””...
  • Page 506 Administrator’s Guide for SIP-T4X 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 507 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 508 <Name>Test2</Name> <Telephone>303</Telephone> <Telephone>915980830849</Telephone> </DirectoryEntry> <DirectoryEntry> <Name>Test3</Name> <Telephone>6650</Telephone> <Telephone>915980830849</Telephone> </DirectoryEntry> </YealinkIPPhoneDirectory> Note Yealink supplies a phone book generation tool to quickly generate a remote XML phone Yealink Phonebook Generation Tool User Guide book. For more information, refer to available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 509 This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using SIP-T4X IP phones. IP phones can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which can help an administrator more easily find the system problem and fix it.
  • Page 510 Administrator’s Guide for SIP-T4X IP Phones Procedure Log setting can be configured using the configuration files or locally. Configures the syslog mode. Parameters: syslog.mode Configures the IP address or domain name of the syslog server where to export the log files.
  • Page 511 Troubleshooting Parameters Permitted Values Default 1-Server Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Settings->Configuration->Export System Log Phone User Interface: None syslog.server IP address or domain name Blank Description: Configures the IP address or domain name of the syslog server when exporting log...
  • Page 512 Administrator’s Guide for SIP-T4X IP Phones Parameters Permitted Values Default Settings->Configuration->System Log Level Phone User Interface: None To configure the system log level via web user interface: Click on Settings->Configuration. Select the desired level from the pull-down list of System Log Level.
  • Page 513 Troubleshooting Enter the IP address or domain name of the syslog server in the Server Name field. 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. Click OK to reboot the phone.
  • Page 514 Administrator’s Guide for SIP-T4X IP Phones The following figure shows a portion of a log file: You can capture packets in two ways: capturing the packets via web user interface or using the Ethernet software. You can analyze the packets captured for troubleshooting purpose.
  • Page 515 Troubleshooting Click Export to open the file download window, and then save the file to your local system. To capture packets using the Ethernet software: Connect the Internet port of the IP phone and the PC to the same HUB, and then use Sniffer, Ethereal or Wireshark software to capture the signal traffic.
  • Page 516 Administrator’s Guide for SIP-T4X IP Phones /servlet?p=settings-prefer ence&q=load Details of the Configuration Parameter: Parameter Permitted Values Default watch_dog.enable 0 or 1 Description : Enables or disables Watch Dog feature. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will reboot automatically when the system is broken down.
  • Page 517 Reading Icons on page 14. This section describes solutions to common issues that may occur while using the IP phone. Upon encountering a scenario not listed in this section, contact your Yealink reseller for further support. Do one of the following: Ensure that the IP phone is properly plugged into a functional AC outlet.
  • Page 518 Administrator’s Guide for SIP-T4X IP Phones Press the OK key when the IP phone is idle to check the basic information (e.g., IP address, MAC address and firmware version). ’ Do one of the following: Ensure that the target firmware is not the same as the current firmware.
  • Page 519 Troubleshooting A remote phone book is placed on a server, while a local phone book is placed on the phone flash. A remote phone book can be used by everyone that can access the server, while a local phone book can only be used by a specific phone. A remote phone book is always used as a central phone book for a company;...
  • Page 520 Administrator’s Guide for SIP-T4X IP Phones Yealink_SIP-T2_Series_T4_Series_IP_Phones_Auto_Provisioning_Guide , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. You can press the volume key to increase or decrease the ringer volume and receiver volume. Press the volume key to adjust the ringer volume when the IP phone is idle, or to adjust the volume of the engaged audio device (handset, speakerphone or headset) when there is an active call in progress.
  • Page 521 Troubleshooting “ ” “ ” They are codes that the IP phone sends to the server when a certain action takes place. On code is used to activate a feature on the server side, while off code is used to deactivate a feature on the server side.
  • Page 522 The <mac>-local.cfg configuration file contains changes made via phone user interface and web user interface. The config.bin file is an encrypte file. For more information on config.bin file, contact your Yealink reseller. To export a BIN configuration file via web user interface:...
  • Page 523 Troubleshooting In the Export or Import Configuration block, click Export to open the file download window, and then save the file to your local system. To export CFG configuration files via web user interface: Click on Settings->Configuration. Select Local Configuration or All Configuration from the pull-down list of Export CFG Configuration File and then click Export to save the file to your local system.
  • Page 524 Administrator’s Guide for SIP-T4X IP Phones In the Export or Import Configuration block, click Browse to locate a BIN configuration file from your local system. Click Import to import the configuration file. To import CFG configuration files via web user interface: Click on Settings->Configuration.
  • Page 525 Appendix 802.1x — an IEEE Standard for port-based Network Access Control (PNAC). It is a part of the IEEE 802.1 group of networking protocols. It offers an authentication mechanism for devices to connect to a LAN or WLAN. ACD (Automatic Call Distribution) — used to distribute calls from large volumes of incoming calls to the registered IP phone users.
  • Page 526 Administrator’s Guide for SIP-T4X IP Phones IEEE (Institute of Electrical and Electronics Engineers) — a non-profit professional association headquartered in New York City that is dedicated to advancing technological innovation and excellence. LAN (Local Area Network) — used to interconnects network devices in a limited area such as a home, school, computer laboratory, or office building.
  • Page 527 Appendix Time Zone Time Zone Name −11:00 Samoa −10:00 United States-Hawaii-Aleutian −09:30 French Polynesia −09:00 United States-Alaska Time −08:00 Canada(Vancouver, Whitehorse) −08:00 Mexico(Tijuana, Mexicali) −08:00 United States-Pacific Time −07:00 Canada(Edmonton, Calgary) −07:00 Mexico(Mazatlan, Chihuahua) −07:00 United States-Mountain Time −07:00 United States-MST no DST −06:00 Canada-Manitoba(Winnipeg) −06:00...
  • Page 528 Administrator’s Guide for SIP-T4X IP Phones Time Zone Time Zone Name Spain-Canary Islands(Las Palmas) United Kingdom(London) Morocco +01:00 Albania(Tirane) +01:00 Austria(Vienna) +01:00 Belgium(Brussels) +01:00 Caicos +01:00 Chad +01:00 Spain(Madrid) +01:00 Croatia(Zagreb) +01:00 Czech Republic(Prague) +01:00 Denmark(Kopenhagen) +01:00 France(Paris) +01:00 Germany(Berlin)
  • Page 529 Appendix Time Zone Time Zone Name +04:00 Russia(Samara) +04:30 Afghanistan(Kabul) +05:00 Kazakhstan(Aqtobe) +05:00 Kyrgyzstan(Bishkek) +05:00 Pakistan(Islamabad) +05:00 Russia(Chelyabinsk) +05:30 India(Calcutta) +05:45 Nepal(Katmandu) +06:00 Kazakhstan(Astana, Almaty) +06:00 Russia(Novosibirsk, Omsk) +06:30 Myanmar(Naypyitaw) +07:00 Russia(Krasnoyarsk) +07:00 Thailand(Bangkok) +08:00 China(Beijing) +08:00 Singapore(Singapore) +08:00 Australia(Perth) +08:00 Russian(Irkutsk, Ulan-Ude) +08:45...
  • Page 530 Administrator’s Guide for SIP-T4X IP Phones This appendix describes the DSS key parameters you can configure on IP phones. DSS keys can be assigned with various key features. The DSS key consists of line key and programable key. The following table lists the number of DSS keys you can configure for...
  • Page 531 Appendix Line    Group Listening  Hot Desking  XML Group  Group Pickup  Multicast Paging  Record  XML Browser  URL Record  LDAP  Prefix  Zero Touch   Local Group  Phone Lock(not applicable to SIP-T48G) ...
  • Page 532 Administrator’s Guide for SIP-T4X IP Phones SIP-T48G/T42G/T41P) Status  Hot Desking (only applicable to  SIP-T48G/T46G) LDAP  Prefix  Local Directory  Local Group  XML Directory  Phone Lock (not applicable to  SIP-T48G) Switch Account Up ...
  • Page 533 Appendix Directory When X=7, the default value is 51 ( Switch Account Up When X=8, the default value is 52 ( Switch Account Down When X=9, the default value is 33 (Status). When X=10, the default value is 0 ( When X=12, the default value is 0 ( When X=13, the default value is 0 ( When X=14, the default value is 2...
  • Page 534 Administrator’s Guide for SIP-T4X IP Phones 10-Call Park 11-DTMF 12-Voice Mail 13-SpeedDial 14-Intercom 15-Line 16-BLF 17-URL 18-Group Listening 22-XML Group 23-Group Pickup 24-Multicast Paging 25-Record 27-XML browser 28-History 30-Menu 32-New SMS (not applicable to SIP-T48G/T42G/T41P) 33-Status 34-Hot Desking 35-URL Record...
  • Page 535 Appendix Parameter- Configuration File linekey.X.line <y0000000000xx>.cfg Parameter- programablekey.X.line Configures the desired line to apply the key feature. For line keys: X ranges from 1 to 29 (for SIP-T48G). X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). For programable keys: X=1-10, 12-14 (for SIP-T48G/T46G) X=1-10, 13 (for SIP-T42G/T41P)
  • Page 536 Administrator’s Guide for SIP-T4X IP Phones  Hot Desking  Phone Lock  Directory  Format Integer For programable keys, the default value is not applicable. For line keys, when x=1, the default value is Default Value When x=2, the default value is 2.
  • Page 537 Appendix Range String within 99 characters When you assign the Speed Dial to the line key, this parameter is used to specify the Example number you want to dial out. linekey.1.value = 1001 Parameter- Configuration File linekey.X.label <y0000000000xx>.cfg Parameter- programablekey.X.label Configures the label displaying on the LCD screen for each line key and each soft key.
  • Page 538 Administrator’s Guide for SIP-T4X IP Phones Example linekey.1.pickup_value = *88 Parameter- Configuration File linekey.X.xml_phonebook <y0000000000xx>.cfg Parameter- programablekey.X.xml_phonebook Configures the desired group or remote phone book when multiple groups or remote phone books are configured on the IP phone. This parameter is only applicable to Local Group/XML Group features.
  • Page 539 Appendix book. linekey.1.xml_phonebook = 1 This section describes how Yealink SIP-T4X IP phones comply with the IETF definition of SIP as described in RFC 3261. This section contains compliance information in the following: RFC and Internet Draft Support  SIP Request ...
  • Page 540 Administrator’s Guide for SIP-T4X IP Phones RFC 3265—Session Initiation Protocol (SIP) - Specific Event Notification  RFC 3266—Support for IPv6 in Session Description Protocol (SDP)  RFC 3310—HTTP Digest Authentication Using Authentication and Key Agreement  (AKA) RFC 3311—The Session Initiation Protocol (SIP) UPDATE Method ...
  • Page 541 Appendix RFC 3863—Presence Information Data Format  RFC 3890—A Transport Independent Bandwidth Modifier for the SDP  RFC 3891—The Session Initiation Protocol (SIP) “Replaces” Header  RFC 3892—The Session Initiation Protocol (SIP) Referred-By Mechanism  RFC 3959—The Early Session Disposition Type for SIP ...
  • Page 542 Administrator’s Guide for SIP-T4X IP Phones and Privacy within Trusted Networks draft-ietf-sipping-cc-conferencing-03.txt—SIP Call Control - Conferencing for User  Agents draft-ietf-sip-connect-reuse-06.txt—Connection Reuse in the Session Initiation  Protocol (SIP) draft-ietf-bliss-shared-appearances-15.txt—Shared Appearances of a Session  Initiation Protocol (SIP) Address of Record (AOR) draft-anil-sipping-bla-04.txt—Implementing Multiple Line Appearances using the...
  • Page 543 Appendix The following SIP request headers are supported: Method Supported Notes Accept Alert-Info Allow Allow-Events Authorization Call-ID Call-Info Contact Content-Length Content-Type CSeq Diversion Event Expires From Max-Forwards Min-SE P-Asserted-Identity P-Preferred-Identity Proxy-Authenticate Proxy-Authorization RAck Record-Route Refer-To Referred-By Remote-Party-ID Replaces...
  • Page 544 Administrator’s Guide for SIP-T4X IP Phones Method Supported Notes Require Route RSeq Session-Expires Subscription-State Supported User-Agent The following SIP responses are supported: 1xx Response—Information Responses 1xx Response Supported Notes 100 Trying 180 Ringing 181 Call Is Being Forwarded 183 Session Progress 2xx Response—Successful Responses...
  • Page 545 Appendix 3xx Response Supported Notes 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 410 Gone...
  • Page 546 Administrator’s Guide for SIP-T4X IP Phones 4xx Response Supported Notes 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...
  • Page 547  The following figure illustrates the scenario of a successful call. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones. The call flow scenario is as follows: User A calls User B.
  • Page 548 Administrator’s Guide for SIP-T4X IP Phones User B hangs up. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 100 Trying F4. 100 Trying F5. 180 Ringing F6. 180 Ringing F7. 200 OK F8. 200 OK F9.
  • Page 549 Appendix Step Action Description The media capability User A is  ready to receive is specified. The port on which User B is  prepared to receive the RTP data is specified. The proxy server maps the SIP URI in the INVITE—Proxy Server to User To field to User B.
  • Page 550 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 551 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 552 The following figure illustrates the scenario of an unsuccessful call caused by the called user’s no answering. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones. The call flow scenario is as follows: User A calls User B.
  • Page 553 Appendix The call cannot be set up successfully. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. CANCEL F6. CANCEL F7. 200 OK F8. 200 OK Step Action Description User A sends an INVITE message to a proxy server.
  • Page 554 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 555 Appendix User A puts User B on hold. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9.
  • Page 556 Administrator’s Guide for SIP-T4X IP Phones Step Action Description The transaction number within a  single call leg is identified in the CSeq field. The media capability User A is  ready to receive is specified. The port on which User B is ...
  • Page 557 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 558 Administrator’s Guide for SIP-T4X IP Phones 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.
  • Page 559 Appendix 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 560 Administrator’s Guide for SIP-T4X IP Phones Step Action Description has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to User B. The ACK confirms that the proxy ACK—Proxy Server to User B server has received the 200 OK response.
  • Page 561 Appendix Step Action Description Server to the proxy server with new SDP session parameters, which are used to place the call on hold. INVITE—Proxy Server to User The proxy server forwards the mid-call INVITE message to User B. User B sends a 200 OK to the proxy 200 OK—User B to Proxy server.
  • Page 562 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 563 Appendix 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. ACK F8. ACK 2-way RTP channel established F9.
  • Page 564 Administrator’s Guide for SIP-T4X IP Phones Step Action Description in the Request-URI field. User A is identified as the call  session initiator in the From field. A unique numeric identifier is  assigned to the call and is inserted in the Call-ID field.
  • Page 565 Appendix Step Action Description User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active. User B sends a REFER message to the REFER—User B to Proxy Server proxy server. User B performs a blind transfer of User A to User C.
  • Page 566 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 567 Appendix 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. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9.
  • Page 568 Administrator’s Guide for SIP-T4X 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 inserted ...
  • Page 569 Appendix 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. The proxy server sends the SIP ACK to User B.
  • Page 570 Administrator’s Guide for SIP-T4X IP Phones 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 INVITE request to User C. User C sends a SIP 180 Ringing 180 Ringing—User C to Proxy...
  • Page 571 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 572 Administrator’s Guide for SIP-T4X IP Phones 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.
  • Page 573 Appendix Step Action Description CSeq field. The media capability User A is  ready to receive is specified. The port on which User B is  prepared to receive the RTP data is specified. The proxy server maps the SIP URI in the INVITE—Proxy Server to User To field to User B.
  • Page 574 User B is busy. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: User B enables busy call forward, and the destination number is User C.
  • Page 575 Appendix 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. 302 Move Temporarily F6. ACK F7. 302 Move Temporarily F8.
  • Page 576 Administrator’s Guide for SIP-T4X IP Phones Step Action Description The media capability User A is  ready to receive is specified. The port on which User B is  prepared to receive the RTP data is specified. The proxy server maps the SIP URI in the INVITE—Proxy Server to User...
  • Page 577 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 578 Administrator’s Guide for SIP-T4X 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. 302 Move Temporarily F6. ACK F7.
  • Page 579 Appendix Step Action Description The media capability User A is  ready to receive is specified. The port on which User B is  prepared to receive the RTP data is specified. The proxy server maps the SIP URI in the INVITE—Proxy Server to User To field to User B.
  • Page 580 User A mixes two RTP channels and therefore establishes a conference between User B and User C. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
  • Page 581 Appendix User A mixes the RTP channels and establishes a conference between User B and User C. User A User B User C Proxy Server F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6.
  • Page 582 Administrator’s Guide for SIP-T4X IP Phones 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.
  • Page 583 Appendix Step Action Description User A sends a mid-call INVITE request INVITE—User A to Proxy to the proxy server with new SDP Server session parameters, which are used to place the call on hold. INVITE—Proxy Server to User The proxy server forwards the mid-call INVITE message to User B.
  • Page 584 Administrator’s Guide for SIP-T4X IP Phones Step Action Description User C is being alerted. User C sends a SIP 200 OK response to 200OK—User C to Proxy the proxy server. The 200 OK response Server notifies User A that the connection has been made.
  • Page 585 Index Numeric 180 Ring Workaround Call Completion 802.1x Authentication Call Display Call Forward Call Hold Call Log About This Guide Call Park Acoustic Echo Cancellation Call Recording Action URL Call Transfer Action URI Call Waiting Administrator Password Calling Line Identification Presentation Always Forward Connected Line Identification Presentation Analyzing the Configuration Files...
  • Page 586 Administrator’s Guide for SIP-T4X IP Phones Enabling the Watch Dog Feature Message Waiting Indicator Missed Call Log Feature Key Synchronization Multicast Paging Music on Hold Getting Information from Status Indicators Getting Started NAT Traversal Group Call Pickup Network Address Translation (NAT)
  • Page 587 Index Server Redundancy Session Timer Wallpaper Web Server Type SIP Components Web User Interface SIP Header SIP Request SIP Responses SIP IP Phone Models SIP Session Description Protocol Usage SIP Session Timer Softkey Layout Specifying the Language to Use SRTP STUN Server Suppressing DTMF Display Table of Contents...

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