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 IP phone firmware contains third-party software under the GNU General Public License (GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms and conditions of the license. The original GPL license, source code of components licensed under GPL and used in Yealink products can be downloaded from Yealink web site: http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293.
Page 5  how to configure BroadSoft features on the BroadWorks web portal and IP phones. For support or service, please contact your Yealink reseller or go to Yealink Technical Support online: http://www.yealink.com/Support.aspx. The information detailed in this guide is applicable to firmware version 73 or higher. The firmware format is like x.x.x.x.rom.
Page 6 RFC 3261, SIP call flows and the sample configuration files. This section describes the changes to this guide for each release and guide version. For more information on changes, refer to version-specific release notes of Yealink IP phones online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
Page 7 About This Guide Notification Popups on page  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-T2xP IP Phones Static DNS Cache on page  Background Noise Suppression on page  Automatic Gain Control on page  Major updates have occurred to the following section: Configuration Files on page  Audio Codecs on page ...
Page 9 About This Guide Resource Files on page  Documentations of the newly released SIP-T19P and SIP-T21P IP phones have also been added. Major updates have occurred to the following sections: Action URL on page  Action URI on page  Major updates have occurred to the following sections: Logo Customization on page...
Page 10 Administrator’s Guide for SIP-T2xP IP Phones The following sections are new for this version: Hot Desking on page  TR-069 Device Management on page  IPv6 Support on page  Major updates have occurred to the following sections: Configuring Network Parameters Manually on page ...
Page 11 About This Guide 180 Ring Workaround on page  Use Outbound Proxy in Dialog on page  SIP Session Timer on page  Session Timer on page  ReCall on page  Transfer via DTMF on page  Intercom on page ...
Page 12 Administrator’s Guide for SIP-T2xP IP Phones Major updates have occurred to the following sections: Dial Plan on page  Phone Lock on page  Time and Date on page  Busy Lamp Field on page ...
Page 13: Table Of Contents
Table of Contents About This Guide ..............v Documentations ..........................v In This Guide ............................ v Summary of Changes ........................vi Changes for Release 73, Guide Version 73.40 ..............vi Changes for Release 73, Guide Version 73.16 ..............vi Changes for Release 72, Guide Version 72.26 ..............vii Changes for Release 72, Guide Version 72.25 ..............
Page 14 Administrator’s Guide for SIP-T2xP IP Phones Provisioning Server ........................20 Supported Provisioning Protocols ..................20 Setting up the Provisioning Server ..................20 Deploying Phones from the Provisioning Server ..............21 Configuring Basic Network Parameters ..................22 DHCP ............................22 Configuring Network Parameters Manually ................ 28 PPPoE ............................
Page 15 Table of Contents Auto Redial ........................... 150 Auto Answer ..........................152 Call Completion ........................... 155 Anonymous Call ........................... 157 Anonymous Call Rejection ......................161 Do Not Disturb ..........................165 Busy Tone Delay ........................... 175 Return Code When Refuse ......................176 Early Media ..........................
Page 16 Administrator’s Guide for SIP-T2xP IP Phones Receiving RTP Stream ......................323 Call Recording ..........................326 Hot Desking ..........................332 Action URL ............................ 335 Action URI ............................. 351 Capturing the Current Screen of the Phone ............... 355 Server Redundancy ........................356 SIP Server Domain Name Resolution ..................
Page 17 Table of Contents Directory Template ........................481 Super Search Template ....................... 482 Local Contact File ........................484 Remote XML Phone Book ......................485 Troubleshooting ..............489 Troubleshooting Methods ......................489 Viewing Log Files ........................489 Capturing Packets ........................ 494 Enabling Watch Dog Feature ....................495 Getting Information from Status Indicators ................
Page 18 Administrator’s Guide for SIP-T2xP IP Phones Appendix E: SIP (Session Initiation Protocol) ................523 RFC and Internet Draft Support ..................523 SIP Request ..........................526 SIP Header ..........................527 SIP Responses ........................528 SIP Session Description Protocol (SDP) Usage ..............531 Appendix F: SIP Call Flows ......................
Page 19: Product Overview
Product Overview This chapter contains the following information about IP phones: VoIP Principle  SIP Components  SIP IP Phone Models  VoIP VoIP (Voice over Internet Protocol) is a technology using the Internet Protocol instead of traditional Public Switch Telephone Network (PSTN) technology for voice communications.
Page 20 Administrator’s Guide for SIP-T2xP IP Phones SIP provides capabilities to: Determine the location of the target endpoint -- SIP supports address resolution,  name mapping, and call redirection. Determine media capabilities of the target endpoint -- Via Session Description ...
Page 21 IP phones comply with the SIP standard (RFC 3261), and they can only be used within a network that supports this model of phone. For a list of key features available on Yealink IP phones running the latest firmware, refer Key Features of IP Phones...
Page 22: Physical Features Of Ip Phones
Administrator’s Guide for SIP-T2xP IP Phones In order to operate as SIP endpoints in your network successfully, IP phones must meet the following requirements: A working IP network is established.  VoIP gateways are configured for SIP .  The latest (or compatible) firmware of IP phones is available.
Page 23 Product Overview SIP-T26P Physical Features: TI TITAN chipset and TI voice engine 132x64 graphic LCD 3 VoIP accounts, Broadsoft Validated/Asterisk Compatible ® HD Voice: HD Codec, HD Handset, HD Speaker 44 keys including 13 DSS keys 1*RJ9 (4P4C) handset port 1*RJ9 (4P4C) headset port 2*RJ45 10/100Mbps Ethernet ports 1*RJ12 (6P6C) expansion module port...
Page 24 Administrator’s Guide for SIP-T2xP IP Phones SIP-T22P Physical Features: TI TITAN chipset and TI voice engine 132x64 graphic LCD 3 VoIP accounts, Broadsoft Validated/Asterisk Compatible ® HD Voice: HD Codec, HD Handset, HD Speaker 31 keys including 3 line keys...
Page 25 Product Overview SIP-T20P Physical Features: TI TITAN chipset and TI voice engine 3-line LCD consists of an icon line and two 15-character lines 2 VoIP accounts, Broadsoft Validated/Asterisk Compatible ® HD Voice: HD Codec, HD Handset, HD Speaker 30 keys including 2 line keys 1*RJ9 (4P4C) handset port 1*RJ9 (4P4C) headset port 2*RJ45 10/100Mbps Ethernet ports...
Page 26 Administrator’s Guide for SIP-T2xP IP Phones In addition to physical features introduced above, IP phones also support the following key features when running the latest firmware: Phone Features  Call Options: emergency call, call waiting, call hold, call mute, call forward, call transfer, call pickup, conference.
Page 27 Product Overview Security  HTTPS (server/client) SRTP (RFC3711) Transport Layer Security (TLS) VLAN (802.1q), QoS Digest authentication using MD5/MD5-sess Secure configuration file via AES encryption Phone lock for personal privacy protection Admin/User configuration mode...
Page 28 Administrator’s Guide for SIP-T2xP IP Phones...
Page 29: Getting Started
Getting Started This chapter provides basic information and installation instructions of IP phones. This chapter provides the following sections: Connecting the IP Phones  Initialization Process Overview  Verifying Startup  Reading Icons  Configuration Methods  Provisioning Server  Configuring Basic Network Parameters ...
Page 30 Administrator’s Guide for SIP-T2xP IP Phones Attach the stand: SIP-T28P/T26P SIP-T22P/T20P Connect the handset and optional headset: SIP-T28P/T26P SIP-T22P/T20P...
Page 31 Getting Started Connect the network and power: AC power (Optional)  Power over Ethernet (PoE)  AC Power (Optional) To connect the AC power and network: Connect the DC plug of the power adapter to the DC5V port on the IP phone and connect the other end of the power adapter into an electrical power outlet.
Page 32 Administrator’s Guide for SIP-T2xP IP Phones To connect the PoE: Connect the Ethernet cable between the Internet port on the IP phone and an available port on the in-line power switch/hub. Note If in-line power switch/hub is provided, you don’t need to connect the phone to the power adapter.
Page 33 Getting Started Querying the DHCP (Dynamic Host Configuration Protocol) Server The IP phone is capable of querying a DHCP server. DHCP is enabled on the IP phone by default. The following network parameters can be obtained from the DHCP server during initialization: IP Address ...
Page 34 Administrator’s Guide for SIP-T2xP IP Phones The message “Initializing… Please Wait” appears on the LCD screen when the IP phone starts up. The main LCD screen displays the following: Time and date  Soft key labels (not applicable to SIP-T20P IP phones) ...
Page 35 Getting Started SIP-T28P SIP-T26P SIP-T22P SIP-T20P Description Call Forward/Forwarded Calls Call Hold Call Mute Ringer volume is 0 Phone Lock Received Calls Placed Calls Missed Calls Recording box is full A call cannot be recorded Recording starts successfully Recording cannot be started Recording cannot be stopped...
Page 36 Access to specific features is restricted to the administrator. The default password is “admin“(case-sensitive). Not all features are available on phone user interface. For more information, refer to Yealink phone-specific user guide, available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. An administrator or a user can configure IP phones via web user interface. The default user name and password for the administrator to log into the web user interface are both “admin”...
Page 37 Scenarios-Protect Personalized Settings Yealink_SIP-T2_Series_T4_Series_IP_Phones_Auto_Provisioning_Guide , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Central Provisioning IP phones can be centrally provisioned from a provisioning server using the configuration files (<y0000000000xx>.cfg and <MAC>.cfg). You can use a text-based editing application to edit configuration files, and then store configuration files to a provisioning server.
Page 38 Administrator’s Guide for SIP-T2xP IP Phones following rules: variable-name = value Associate only one value with one variable. Separate each variable name and value with an equal sign. Set only one variable per line. Put the variable and value on the same line, and do not break the line.
Page 39 MAC-oriented configuration file will override the same one in the common configuration file. Yealink supplies configuration files for each phone model, which is delivered with the phone firmware. The configuration files, supplied with each firmware release, must be used with that release.
Page 40 Administrator’s Guide for SIP-T2xP IP Phones IP phones discover the provisioning server address, and then download the configuration files from the provisioning server. For more information on configuration files, refer to Configuration Files on page 18. For more information on encrypting configuration files, refer to...
Page 41 Getting Started DHCP Option DHCP provides a framework for passing information to TCP/IP network devices. Network and other control information are carried in tagged data items that are stored in the options field of the DHCP message. The data items themselves are also called options. DHCP can be initiated by simply connecting the IP phone with the network.
Page 42 Administrator’s Guide for SIP-T2xP IP Phones Parameter DHCP Option Description Identify a boot file when the 'file' field in the Boot file Name DHCP header has been used for DHCP options. For more information on DHCP options, refer to http://www.ietf.org/rfc/rfc2131.txt?number=2131 http://www.ietf.org/rfc/rfc2132.txt?number=2132.
Page 43 Getting Started 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. 0-DHCP 1-PPPoE 2-Static IP Address Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 44 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Example: network.primary_dns = 202.101.103.55 Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Basic->IPv4 Config->Static IP Address->Primary DNS Phone User Interface: Menu->Settings->Advanced Settings (default password: admin) ->Network->WAN...
Page 45 Getting Started In the IPv4 Config block, mark the DHCP radio box. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone. To configure static DNS address when DHCP is used via web user interface: Click on Network->Basic.
Page 46 Administrator’s Guide for SIP-T2xP IP Phones To configure DHCP via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->WAN Port->IPv4. Press to highlight the DHCP IPv4 Client field. The IP phone reboots automatically to make settings effective after a period of time.
Page 47 Getting Started Configure network parameters of the IP phone manually. Web User Interface Navigate to: Local http://<phoneIPAddress>/servlet ?p=network&q=load Configure network parameters of Phone User Interface the IP phone manually. 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 48 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Port->IP Mode network.internet_port.ip IPv4 Address Blank Description: Configures the IPv4 address when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP Address.
Page 49 Getting Started Parameters Permitted Values Default Description: Configures the IPv4 default gateway when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP Address. Example: network.internet_port.gateway = 192.168.1.254 Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 50 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Configures the secondary IPv4 DNS server when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP Address.
Page 51 Getting Started Enter the desired values in the IP Address, Subnet Mask, Gateway, 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 52 Administrator’s Guide for SIP-T2xP IP Phones Contact your ISP for the PPPoE user name and password. Procedure PPPoE can be configured using the configuration files or locally. Configure PPPoE on the IP phone. <MAC>.cfg Parameters: network.internet_port.type Configure the user name and...
Page 53 Getting Started Parameters Permitted Values Default String within 32 network.pppoe.user Blank characters Description: Configures the user name for PPPoE connection when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet port type is configured as PPPoE. Example: network.pppoe.user = xmyealink Note: If you change this parameter, the IP phone will reboot to make the change take...
Page 54 Administrator’s Guide for SIP-T2xP IP Phones Enter the user name and password in corresponding 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 55 Getting Started transmission. Half-duplex Half-duplex transmission refers to transmitting voice or data in both directions, but in one direction at a time; this means one device can send data on the line, but not receive data simultaneously. You can configure the half-duplex transmission on both Internet port and PC port for the IP phone to transmit in 10Mbps or 100Mbps.
Page 56 Administrator’s Guide for SIP-T2xP IP Phones Configure the transmission methods of Ethernet ports. Local Web User Interface Navigate to: http://<phoneIPAddress>/servlet?p= network-adv&q=load Details of Configuration Parameters: Parameters Permitted Values Default network.internet_port.speed_duplex 0, 1, 2, 3 or 4 Description: Configures the transmission method and speed of the Internet (WAN) port.
Page 57 Getting Started Parameters Permitted Values Default Phone User Interface: None To configure the transmission methods of Ethernet ports via web user interface: Click on Network->Advanced. Select the desired value from the pull-down list of WAN Port Link. Select the desired value from the pull-down list of PC Port Link. Click Confirm to accept the change.
Page 58 Administrator’s Guide for SIP-T2xP IP Phones ?p=network-pcport&q=load Details of Configuration Parameters: Parameters Permitted Values Default network.PC_port.enable 0 or 1 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.
Page 59: Upgrading Firmware
9.73.0.40.rom Note You can download the latest firmware online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Do not unplug the network and power cables when the IP phone is upgrading firmware. Upgrade via Web User Interface To manually upgrade firmware via web user interface, you need to store firmware to your local system in advance.
Page 60 Administrator’s Guide for SIP-T2xP IP Phones Click Browse. Select firmware from the local system. Click Upgrade. 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!”.
Page 61 Getting Started auto_provision.weekly.begin_time 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=s ettings-autop&q=load Details of Configuration Parameters: Parameters Permitted Values Default auto_provision.power_on 0 or 1...
Page 62 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default None 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 parameter “auto_provision.repeat.enable” is set to 1(Enabled).
Page 63 Getting Started 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 parameter “auto_provision.weekly.enable” is set to 1(Enabled). Web User Interface: Settings->Auto provision->Time Phone User Interface: None...
Page 64 Administrator’s Guide for SIP-T2xP IP Phones 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.
Page 65 Getting Started 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 66 Administrator’s Guide for SIP-T2xP IP Phones...
Page 67: Notification Popups
Configuring Basic Features This chapter provides information for making configuration changes for the following basic features: Power Indicator LED  Notification Popups  Contrast  Backlight  Call Display  Web Server Type  User Password  Administrator Password  Phone Lock ...
Page 68: Early Media
Administrator’s Guide for SIP-T2xP IP Phones Do Not Disturb  Busy Tone Delay  Return Code When Refuse  Early Media  180 Ring Workaround  Use Outbound Proxy in Dialog  SIP Session Timer  Session Timer  Call Hold ...
Page 69 Configuring Basic Features Voice/Text Mail Power Light Flash Voice/Text Mail Power Light Flash allows the power indicator LED to flash when the IP phone receives a voice mail or a text message. Mute Power Light Flash Mute Power Light Flash allows the power indicator LED to flash when a call is mute. Hold/Held Power Light Flash Hold/Held Power Light Flash allows the power indicator LED to flash when a call is placed on hold or is held.
Page 70 Administrator’s Guide for SIP-T2xP IP Phones Details of Configuration Parameters: Permitted Parameters Default Values phone_setting.common_power_led_enable 0 or 1 Description: Enables or disables the power indicator LED to be turned on. 0-Disabled (power indicator LED is off) 1-Enabled (power indicator LED is solid green) Note: The old parameter “features.power_led_on”...
Page 71 Configuring Basic Features Permitted Parameters Default Values 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) green) Web User Interface: Features->Power LED->Mute Power Light Flash Phone User Interface:...
Page 72 Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of Ringing Power Light Flash Select the desired value from the pull-down list of Voice/Text Mail Power Light Flash. Select the desired value from the pull-down list of Mute Power Light Flash.
Page 73 Configuring Basic Features Navigate to: http://<phoneIPAddress>/servlet?p=f eatures-notifypop&q=load Details of Configuration Parameters: Permitted Parameters Default Values features.voice_mail_popup.enable 0 or 1 Description: Enables or disables the IP phone to display the pop-up message box when it receives a new voice mail. 0-Disabled 1-Enabled Note: If the voice mail pop-up message box disappears, it won't pop up again unless the user receives a new voice mail or the user re-registers the account that has unread...
Page 74 Administrator’s Guide for SIP-T2xP IP Phones Permitted Parameters Default Values 0-Disabled 1-Enabled Note: It is only applicable to IP phones running firmware version 73 or later. Web User Interface: Features->Notification Popups->Display Forward Call Popup Phone User Interface: None features.text_message_popup.enable 0 or 1...
Page 75 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 76 Administrator’s Guide for SIP-T2xP IP Phones Details of the Configuration Parameter: Parameter Permitted Values Default phone_setting.contrast Integer from 1 to 10 Description: Configures the contrast of the LCD screen. For SIP-T28P IP phones, it configures the LCD’s contrast of the IP phone and the connected EXP39.
Page 77: Configuration Methods
Configuring Basic Features Press the Save soft key to accept the change. Note Before you adjust the LCD’s contrast of the expansion module, make sure the expansion module has been connected to the IP phone. Backlight determines the brightness of the LCD screen display, allowing users to read easily in dark environments.
Page 78 Administrator’s Guide for SIP-T2xP IP Phones Phone Model Configuration Methods Configuration Options Web User Interface Phone User Interface Procedure Backlight can be configured using the configuration files or locally. Configure the backlight of the LCD screen. Configuration Parameters: <y0000000000xx>.cfg File phone_setting.active_backlight_level...
Page 79 Configuring Basic Features Parameters Permitted Values Default when the IP phone is inactive. 0-Always on 1-Always off 15-15s 30-30s 60-60s 120-120s 300-300s 600-600s 1800-1800s If it is set to 60 (60s), the intensity of the LCD screen will be changed when the IP phone is inactive for 60 seconds.
Page 80 Administrator’s Guide for SIP-T2xP IP Phones To configure backlight via phone user interface (only applicable to SIP-T28P IP phones and EXP39 connected to SIP-T26P and SIP-T28P IP phones): Press Menu->Settings->Basic Settings->Display->Backlight. Press , or the Switch soft key to select the desired level from the Backlight Level field.
Page 81 Configuring Basic Features Procedure Web server type can be configured using the configuration files or locally. Enable or disable display called party information feature. Parameter: phone_setting.called_party_inf o_display.enable Configuration File <y0000000000xx>.cfg Sepecify the type of caller information display. Parameter: phone_setting.call_info_display _method Configure call display features.
Page 82 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Specifies the caller and callee information display method when the IP phone receives an incoming call, dials an outgoing call or is during an active call 0-Name+Number 1-Number+Name 2-Name 3-Number 4-Full Contact Info (display name<sip:xxx@domain.com>)
Page 83 Configuring Basic Features a web protocol that encrypts and decrypts user page requests as well as pages returned by the web server. Both HTTP and HTTPS port numbers are configurable. Procedure Web server type can be configured using the configuration files or locally. Configure the web access type, HTTP port and HTTPS port.
Page 84 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default network.port.http Integer from 1 to 65535 Description: Configures the HTTP port for the user to access web user interface of the IP phone using the HTTP protocol. The default HTTP port is 80.
Page 85 Configuring Basic Features Parameters Permitted Values Default Description: Configures the HTTPS port for the user to access web user interface of the IP phone using the HTTPS protocol. The default HTTPS port is 443. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 86 Administrator’s Guide for SIP-T2xP IP Phones Click OK to reboot the phone. To configure web server type via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->Webserver Type. Press , or the Switch soft key to select the desired value from the HTTP Status field.
Page 87 Configuring Basic Features Parameter Permitted Values Default Description: Configures the password of the user for web server access. The IP phone uses “user” as the default user password. The valid value format is username:new password. Example: security.user_password = user:password123 means setting the password of user (current user name is “user”) to password123.
Page 88 Administrator’s Guide for SIP-T2xP IP Phones changed by an administrator. The default administrator password is “admin”. For security reasons, the administrator should change the default administrator password as soon as possible. Procedure Administrator password can be changed using the configuration files or locally.
Page 89 Configuring Basic Features Enter the current administrator password in the Old Password field. Enter new password in the New Password and Confirm Password fields. Valid characters are ASCII characters 32-126(0x20-0x7E) except 58(3A). Click Confirm to accept the change. To change the administrator password via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Set Password.
Page 90: Phone User Interface
Administrator’s Guide for SIP-T2xP IP Phones Change the unlock PIN. Parameter: phone_setting.phone_lock.unlock_pin Configure the IP phone to automatically lock the keypad after a time interval. Parameter: phone_setting.phone_lock.lock_time_out Assign a phone lock key. Parameter: memorykey.X.type/linekey.X.type/ programablekey.X.type Configure the type of phone lock.
Page 91 Configuring Basic Features Parameters Permitted Values Default key is locked). Function Keys: MESSAGE, RD, CONF , HOLD, MUTE, TRAN, OK, X, navigation keys, soft keys, line keys and memory keys are locked (For SIP-T22P , CONF , HOLD, MUTE and memory keys do not exist; For SIP-T20P , the MUTE key, soft keys and memory keys do not exist, but the additional MENU and Directory keys are locked).
Page 92 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default None Phone Lock Key For more information on how to configure the DSS Key, refer to Appendix D: Configuring DSS Key on page 513. Parameter Permitted Values Default memorykey.X.type/ linekey.X.type/ Refer to the programablekey.X.type...
Page 93 Configuring Basic Features Parameter Permitted Values Default When X=4, the default value is 30 (Menu). When X=5, the default value is 28 ( History When X=6, the default value is 61 ( Directory When X=7, the default value is 31 ( Switch Account When X=8, the default value is 31 ( Switch Account...
Page 94 Administrator’s Guide for SIP-T2xP IP Phones Parameter Permitted Values Default Phone User Interface: Menu->Features->DSS Keys->Memory Keys (or Line Keys)->DSS Key X (or Line Key X)->Type To configure phone lock via web user interface: Click on Features->Phone Lock. Select the desired type from the pull-down list of Phone Lock Type.
Page 95: Time Zone
Configuring Basic Features Click Confirm to accept the change. To configure the type of phone lock via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Phone Lock. Press , or the Switch soft key to select the desired type from the Phone Lock field.
Page 96: Configuration Files
Administrator’s Guide for SIP-T2xP IP Phones clocks are adjusted forward one hour at the start of spring and backward in autumn. Many countries have used the DST at various times, details vary by location. The DST can be adjusted automatically from the time zone configuration. Typically, there is no need to change this setting.
Page 97 Configuring Basic Features local_time.time_zone_name local_time.summer_time local_time.dst_time_type local_time.start_time local_time.end_time local_time.offset_time Configure the time and date manually. Parameter: local_time.manual_time_enable Configure the time and date formats. Parameters: local_time.time_format local_time.date_format Configure NTP by DHCP priority feature. Configure the NTP server, time zone and DST. Configure the time and date Web User Interface manually.
Page 98 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Enables or disables the IP phone to use manually configured NTP server preferentially. 0-High (use the NTP server obtained by DHCP preferentially) 1-Low (use the NTP server configured manually preferentially) Web User Interface: Settings->Time &...
Page 99 Configuring Basic Features Parameters Permitted Values Default Configures the IP address or the domain name of the NTP server 2. If the NTP server 1 is not configured or cannot be accessed, the IP phone will request the time and date from the NTP server 2.
Page 100 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default parameter “local_time.time_zone”. For more information on the available time zone names for each time zone, refer to Appendix B: Time Zones on page 509. Note: It works only if the value of the parameter “local_time.summer_time” is set to 2 (Automatic).
Page 101 Configuring Basic Features Parameters Permitted Values Default Configures the start time of the DST. Value formats are: Month/Day/Hour (for By Date)  Month/ Day of Week Last in Month/ Day of Week/ Hour of Day (for By Week)  If “local_time.dst_time_type” is set to 0 (By Date), use the mapping: Month: 1=Jan, 2=Feb,…, 12=Dec Day:1=the first day in a month,…, 31= the last day in a month Hour:0=0am, 1=1am,…, 23=11pm...
Page 102 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default If “local_time.dst_time_type” is set to 1 (By Week), use the mapping: Month: 1=Jan, 2=Feb,…, 12=Dec Day of Week Last in Month: 1=the first week in a month,…, 5=the last week in a month Day of Week: 1=Mon, 2=Tues,…, 7=Sun...
Page 103 Configuring Basic Features Parameters Permitted Values Default local_time.time_format 0 or 1 Description: Configures the time format. 0-12 Hour 1-24 Hour If it is set to 0 (12 Hour), the time will be displayed in 12-hour format with AM or PM specified.
Page 104 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default For SIP-T22P/T26P/T28P IP phones, the default value is 0. For SIP-T20P IP phones, the default value is 7. Note: “WWW” represents the abbreviation of the week, “DD” represents a two-digit day, “MMM”...
Page 105 Configuring Basic Features If you select Enabled, do one of the following: Mark the DST By Date radio box in the Fixed Type field. Enter the start time in the Start Date field. Enter the end time in the End Date field. Mark the DST By Week radio box in the Fixed Type field.
Page 106 Administrator’s Guide for SIP-T2xP IP Phones Click Confirm to accept the change. To configure the time and date manually via web user interface: Click on Settings->Time & Date. Select Enabled from the pull-down list of Manual Time. Enter the time and date in the corresponding fields.
Page 107 Configuring Basic Features The default time zone is "+8 China(Beijing)". Enter the domain names or IP addresses in the NTP Server1 and NTP Server2 fields respectively. Press the Save soft key to accept the change. To configure the time and date manually via phone user interface: Press Menu->Settings->Basic Settings->Time &...
Page 108 Administrator’s Guide for SIP-T2xP IP Phones Phone User Interface Web User Interface Russian Languages available for selection depend on language packs currently loaded to the IP phone. You can customize the translation of the existing language on the phone user interface or web user interface. You can also make new languages available for use on the phone user interface and web user interface by loading language packs to the IP phone.
Page 109 Configuring Basic Features To customize a language file: Open the desired language template file (e.g., 000.GUI.English.lang) using an ASCII editor. Modify the characters within the double quotation marks on the right of the equal sign. Don’t modify the translation item on the left of the equal sign. The following shows a portion of the language pack “000.GUI.English.lang”...
Page 110 Administrator’s Guide for SIP-T2xP IP Phones Associated Language Associated Note Available Language Pack Language Pack Italian 6.Italian.js 6.Italian_note.xml Polish 7.Polish.js 7.Polish_note.xml Portuguese 8.Portuguese.js 8.Portuguese_note.xml Spanish 9.Spanish.js 9.Spanish_note.xml Turkish 10.Turkish.js 10.Turkish_note.xml Russian 11.Russian.js 11.Russian_note.xml When adding a new language pack for the web user interface, the language pack must be formatted as “Y.name.js”...
Page 111 Configuring Basic Features pack). To customize a note language file: Open the desired note language template file (e.g., 1.English_note.xml) using an ASCII editor. Modify the text of the note field. Don't modify the name of the note field. The following shows a portion of the note language pack “1.English_note.xml” for the web user interface: Note The new added language must be supported by the font library on the IP phone.
Page 112 Administrator’s Guide for SIP-T2xP IP Phones wui_lang.url Specify the access URL of the note language pack of the web user interface Parameter: wui_lang_note.url Delete customized language packs of the phone user interface Parameter: gui_lang.delete Delete customized language packs and note language packs of the web user interface.
Page 113 Configuring Basic Features Parameter Permitted Values Default None http://localhost/all or X.GUI.nam gui_lang.delete http://localhost/ Blank e.lang Description: Deletes the specified or all customized language packs of the phone user interface. Example: Delete all customized language packs of the phone user interface. gui_lang.delete = http://localhost/all Delete a customized language pack of the phone user interface (e.g., 008.GUI.Russian.lang)
Page 114 Administrator’s Guide for SIP-T2xP IP Phones Parameter Permitted Values Default None URL within 511 characters Blank wui_lang_note.url Description: Configures the access URL of the language pack for web note. Example: The following example uses HTTP to download the language pack “1.English_note.xml”...
Page 115 Configuring Basic Features Parameter Permitted Values Default deleted. Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is only applicable to IP phones running firmware version 73 or later. Web User Interface: None Phone User Interface: None...
Page 116 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Configures the language used on the phone user interface. Permitted Values: English, Chinese_S, Chinese_T, French, German, Italian, Portuguese, Polish, Spanish, Turkish, Russian or the custom language name. Example: lang.gui = English...
Page 117 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 118 Administrator’s Guide for SIP-T2xP IP Phones The following shows a portion of the input method file “ime.txt”:...
Page 119 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 120 Administrator’s Guide for SIP-T2xP 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 121 Before uploading your custom logo to IP phones, ensure your logo file is correctly formatted. For more information on customizing a logo file, refer to Yealink_SIP-T2_Series_T4_Series_IP_Phones_Auto_Provisioning_Guide , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Procedure The logo shown on the idle screen can be configured using the configuration files or locally.
Page 122 Administrator’s Guide for SIP-T2xP IP Phones ?p=features-general&q=load Details of Configuration Parameters: Parameters Permitted Values Default Refer to the following phone_setting.lcd_logo.mode 0, 1 or 2 content 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 123 Yealink characters Description: Configures a text logo. Example: phone_setting.lcd_logo.text = Yealink Note: It is only applicable to SIP-T20P IP phones. Web User Interface: Features->General Information->Text Logo Phone User Interface: None To configure an image logo via web user interface (not applicable to SIP-T20P IP phones): Click on Features->General Information.
Page 124 Administrator’s Guide for SIP-T2xP IP Phones 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. For SIP-T28P IP phones, the image logo is displayed on the idle screen. For SIP-T26P/T22P IP phones, the image logo screen and the idle screen are displayed alternately.
Page 125 Configuring Basic Features Enter the desired text (0~15 characters) in the Text Logo field. Click Confirm to accept the change. The registered account and the configured text logo are displayed alternately. Softkey layout is used to customize the soft keys at the bottom of the LCD screen to best meet users’...
Page 126 Administrator’s Guide for SIP-T2xP IP Phones Call State Default Soft Keys Optional Soft Keys Empty Empty Empty Switch Connecting Empty Cancel Connecting Transfer Empty Empty Switch SemiAttendTrans Empty Cancel Send Empty History Delete Switch Cancel Line Dialing Directory GPickup DPickup...
Page 127 Configuring Basic Features Call State Default Soft Keys Optional Soft Keys Resume Switch NewCall Answer Cancel Reject Empty Empty Empty Switch Held Empty Answer Cancel Reject NewCall Transfer Empty Directory PreTrans Delete Switch Cancel Send Empty Empty Hold Switch Conferenced Split Answer Cancel...
Page 128 Administrator’s Guide for SIP-T2xP IP Phones Details of Configuration Parameters: Parameters Permitted Values Default phone_setting.custom_softkey_enable 0 or 1 Description: Enables or disables custom soft keys layout feature. 0-Disabled 1-Enabled Web User Interface: Settings->Softkey Layout->Custom Softkey Phone User Interface: None custom_softkey_call_failed.url...
Page 129 Configuring Basic Features Parameters Permitted Values Default Phone User Interface: None custom_softkey_connecting.url URL within 511 characters Blank Description: Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Connecting state. Example: The following example uses HTTP to download the Connecting state file from the “XMLfiles”...
Page 130 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_ring_back.url = http://10.2.8.16:8080/XMLfiles/RingBack.xml Web User Interface: None Phone User Interface: None custom_softkey_talking.url URL within 511 characters Blank Description: Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Talking state.
Page 131 Configuring Basic Features To adjust the display order of soft keys, select the desired soft key and then click The LCD screen displays the soft keys in the adjusted order. Click Confirm to accept the change. Key as send allows assigning the pound key or asterisk key as a send key. Send sound allows the IP phone to play a key tone when a user presses the send key.
Page 132 Administrator’s Guide for SIP-T2xP IP Phones ?p=features-general&q=load Configure a send sound and key tone. Navigate to: http://<phoneIPAddress>/servlet ?p=features-audio&q=load Configure the send key. Phone User Interface Configure a key tone. Details of Configuration Parameters: Parameters Permitted Values Default features.key_as_send 0, 1 or 2 Description: Configures the "#"...
Page 133 Configuring Basic Features Parameters Permitted Values Default Features->Audio->Key Tone Phone User Interface: Menu->Settings->Basic Settings->Sound->Key Tone features.send_key_tone 0 or 1 Description: Enables or disables the IP phone to play a tone when a user presses a send key. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will play a tone when a user presses a send key.
Page 134 Administrator’s Guide for SIP-T2xP IP Phones To configure a send sound and key tone via web user interface: Click on Features->Audio. Select the desired value from the pull-down list of Key Sound. Select the desired value from the pull-down list of Send Sound.
Page 135 Configuring Basic Features Dial-now  Area Code  Block Out  You need to know the following basic regular expression syntax when creating dial plan: The dot “.” can be used as a placeholder or multiple placeholders for any string. Example: “12.”...
Page 136 Administrator’s Guide for SIP-T2xP IP Phones Procedure Replace rule can be created using the configuration files or locally. Create the replace rule for the IP phone. Parameters: dialplan.replace.prefix.X dialplan.replace.replace.X Configuration File <y0000000000xx>.cfg dialplan.replace.line_id.X Configure the access URL of the replace rule template.
Page 137 Configuring Basic Features Parameters Permitted Values Default Description: Configures the alternate number to replace the entered number. Example: dialplan.replace.replace.1 = 123456 Web User Interface: Settings->Dial Plan->Replace Rule->Replace Phone User Interface: None dialplan.replace.line_id.X Blank (for Refer to the following content all lines) (X ranges from 1 to 100) Description: Configures the desired line to apply the replace rule.
Page 138 Administrator’s Guide for SIP-T2xP IP Phones To create a replace rule via web user interface: Click on Settings->Dial Plan->Replace Rule. Enter the string in the Prefix field. Enter the string in the Replace field. Enter the desired line ID in the Account field or leave it blank.
Page 139 Configuring Basic Features Procedure Dial-now rule can be created using the configuration files or locally. Create the dial-now rule for the IP phone. Parameters: dialplan.dialnow.rule.X dialplan.dialnow.line_id.X Configuration File <y0000000000xx>.cfg Configure the delay time for the dial-now rule and the access URL of the dial-now template.
Page 140 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Settings->Dial Plan->Dial-now->Rule Phone User Interface: None dialplan.dialnow.line_id.X Blank (for Refer to the following content all lines) (X ranges from 1 to 100) Description: Configures the desired line to apply the dial-now rule. The digit 0 stands for all lines.
Page 141 Configuring Basic Features Parameters Permitted Values Default Description: Configures the access URL of the dial-now rule template file. Example: dialplan_dialnow.url = http://192.168.10.25/dialnow.xml Web User Interface: None Phone User Interface: None To create a dial-now rule via web user interface: Click on Settings->Dial Plan->Dial-now. Enter the desired value in the Rule field.
Page 142 Administrator’s Guide for SIP-T2xP IP Phones Enter the desired time within 1-14 (in seconds) in the Time-Out for Dial-Now Rule field. Click Confirm to accept the change. Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country.
Page 143 Configuring Basic Features numbers. Navigate to: http://<phoneIPAddress>/servlet ?p=settings-areacode&q=load Details of Configuration Parameters: Parameters Permitted Values Default dialplan.area_code.code String within 16 characters Blank Description: Configures the area code to be added before the entered numbers when dialing out. Note: The length of the entered number must be between the minimum length configured by the parameter “dialplan.area_code.min_len”...
Page 144 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Phone User Interface: None Blank (for dialplan.area_code.line_id Refer to the following content all lines) Description: Configures the desired line to apply the area code rule. The digit 0 stands for all lines.
Page 145 Configuring Basic Features If you leave this field blank or enter 0, the area code rule will apply to all accounts on the IP phone. Click Confirm to accept the change. Block out rule prevents users from dialing out specific numbers. When entered numbers match the predefined block out rule, the LCD screen prompts “Forbidden Number”.
Page 146 Administrator’s Guide for SIP-T2xP IP Phones 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 = 5432 Web User Interface: Settings->Dial Plan->Block Out->BlockOut NumberX...
Page 147 Configuring Basic Features If you leave this 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 148 Administrator’s Guide for SIP-T2xP IP Phones number. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Configure the hotline number. Specify the time (in seconds) the Phone User Interface IP phone waits before automatically dialing out the hotline number. Details of Configuration Parameters: Parameter Permitted Values Default features.hotline_number...
Page 149 Configuring Basic Features Parameter Permitted Values Default Menu->Features->Hotline->Hotline Delay To configure hotline via web user interface: Click on Features->General Information. Enter the hotline number in the Hotline Number field. Enter the delay time in the Hotline Delay (0~10s) field. Click Confirm to accept the change. To configure hotline via phone user interface: Press Menu->Features->Hot Line.
Page 150 Administrator’s Guide for SIP-T2xP IP Phones Off hook hot line dialing feature is configurable on a per-line basis and depends on support from a SIP server. Note Off hook hot line dialing feature limits the call-out permission of this account and disables the hotline feature.
Page 151 Configuring Basic Features Parameter Permitted Values Default Web User Interface: None Phone User Interface: None account.X.auto_dial_num String within 32 characters Blank Description: Configures the number that the IP phone first dials out when a user presses the speakerphone key or desired line key, dials out a call or off hook the phone using account X.
Page 152 Administrator’s Guide for SIP-T2xP IP Phones http://<phoneIPAddress>/servlet ?p=contacts-favorite&q=load Details of the Configuration Parameter: Parameter Permitted Values Default directory_setting.url URL within 511 characters Blank 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...
Page 153 Configuring Basic Features Click Confirm to accept the change. The IP phone LCD screen will display the enabled list(s) in the adjusted order. 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.
Page 154 Administrator’s Guide for SIP-T2xP IP Phones Parameter Permitted Values Default super_search.url URL within 511 characters Blank Description: Configures the access URL of the super search template. Web User Interface: Directory->Setting->Search Source List In Dialing Phone User Interface: None To configure search source list in dialing via web user interface: Click on Directory->Setting.
Page 155 Configuring Basic Features Call log contains call information such as remote party identification, time and date, and call duration. It can be used to redial previous outgoing calls, return incoming calls, and save contact information from call log lists to the contact directory. IP phones maintain a local call log.
Page 156 Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of Save Call Log. Click Confirm to accept the change. To configure call log feature via phone user interface: Press Menu->Features->History Setting. Press , or the Switch soft key to select the desired value from the History Record field.
Page 157 Configuring Basic Features Details of the Configuration Parameter: Parameter Permitted Values Default account.X.missed_calllog 0 or 1 Description: Enables or disables the IP phone to record missed calls for account X. 0-Disabled 1-Enabled If it is set to 0 (Disabled), there is no indicator displaying on the LCD screen, the IP phone does not log the missed call in the Missed Calls list.
Page 158 Contacts and groups can be added either one by one or in batch using a local contact file. Yealink IP phones support both *.xml and *.csv format contact files. For more information on how to customize a contact file (*.xml), refer to Local Contact File page 484.
Page 159 Configuring Basic Features Add a group and a contact to the Phone User Interface local directory. Details of the Configuration Parameter: Parameter Permitted Values Default local_contact.data.url URL within 511 characters Blank Description: Configures the access URL of the local contact file (*.xml). Example: local_contact.data.url = http://192.168.10.25/contact.xml Web User Interface:...
Page 160 Administrator’s Guide for SIP-T2xP IP Phones To add a contact to the local directory via web user interface: Click on Directory->Local Directory. In the Directory block, enter the name and the office, mobile or other numbers in the corresponding fields.
Page 161 Configuring Basic Features Click Browse to locate a contact list file (the file format must be *.xml) from your local system. Click Import XML to import the contact list. The web user interface prompts "The original contact will be covered, Continue?". Click OK to complete importing the contact list.
Page 162 Administrator’s Guide for SIP-T2xP IP Phones Select the contact information you want to import into the local directory from the pull down list of Index. At least one row information should be selected to be imported into the local directory.
Page 163 Configuring Basic Features Live dialpad allows IP phones to automatically dial out the entered phone number after a specified period of time. Procedure Live dialpad can be configured using the configuration files or locally. Configure live dialpad. Parameters: Configuration File <y0000000000xx>.cfg phone_setting.predial_autodial phone_setting.inter_digit_time...
Page 164 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default (Enabled). Web User Interface: Settings->Preference->Inter Digit Time (1~14s) Phone User Interface: None To configure live dialpad via web user interface: Click on Settings->Preference. Select the desired value from the pull-down list of Live Dialpad.
Page 165 Configuring Basic Features Procedure Call waiting and call waiting tone can be configured using the configuration files or locally. Configure call waiting and call waiting tone. Parameters: call_waiting.enable Configuration File <y0000000000xx>.cfg call_waiting.tone call_waiting.on_code call_waiting.off_code Configure call waiting. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Web User Interface Configure call waiting tone.
Page 166 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Menu->Features->Call Waiting->Call Waiting call_waiting.tone 0 or 1 Description: Enables or disables the IP phone to play the call waiting tone when the IP phone receives an incoming call during a call...
Page 167 Configuring Basic Features Parameters Permitted Values Default Web User Interface: Features->General Information->Call Waiting Off Code Phone User Interface: Menu->Features->Call Waiting->Off Code To configure call waiting via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Call Waiting. (Optional.) Enter the call waiting on code in the Call Waiting On Code field.
Page 168 Administrator’s Guide for SIP-T2xP IP Phones 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 169 Configuring Basic Features Parameters Permitted Values Default 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 170 Administrator’s Guide for SIP-T2xP IP Phones Enter the waiting time in the Auto Redial Interval (1~300s) field. The default waiting time is 10s. Enter the desired times in the Auto Redial Times (1~300) field. The default value is 10. Click Confirm to accept the change.
Page 171 Configuring Basic Features 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. Navigate to: http://<phoneIPAddress>/servlet ?p=account-basic&q=load&acc Web User Interface...
Page 172 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Account->Basic->Auto Answer Phone User Interface: Menu->Settings->Advanced Settings (default password: admin)->Account->Account X->Auto Answer features.auto_answer_delay Integer from 1 to 4 (X ranges from 1 to 6) Description: Configures the delay time (in seconds) before the IP phone automatically answers an incoming call.
Page 173 Configuring Basic Features Enter the desired time in the Auto-Answer Delay (1~4s) field. Click Confirm to accept the change. To configure auto answer via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Accounts. Select the desired account and then press the Enter soft key. Press , or the Switch soft key to select the desired value from the Auto Answer field.
Page 174 Administrator’s Guide for SIP-T2xP IP Phones 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.
Page 175 Example of anonymous SIP header: Via: SIP/2.0/UDP 10.2.8.183:5063;branch=z9hG4bK1535948896 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=128043702 To: <sip:1011@10.2.1.199> Call-ID: 1773251036@10.2.8.183 CSeq: 1 INVITE Contact: <sip:1012@10.2.8.183:5063> Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T28P 2.72.0.1...
Page 176 Administrator’s Guide for SIP-T2xP IP Phones 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 177 Configuring Basic Features Parameters Permitted Values Default X ranges from 1 to 3 (for SIP-T26P/T22P). X ranges from 1 to 2 (for SIP-T20P). Web User Interface: Account->Basic->Local Anonymous Phone User Interface: Menu->Features->Anonymous Call->Local Anonymous account.X.send_anonymous_code 0 or 1 Description: Configures the IP phone to send anonymous on/off code to activate/deactivate the server-side anonymous call feature for account X.
Page 178 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default (On Code). Web User Interface: Account->Basic->Anonymous Call->On Code Phone User Interface: Menu->Features->Anonymous Call->Send Anony Code->On Code account.X.anonymous_call_offcode String within 32 characters Blank Description: Configures the anonymous call off code to deactivate the server-side anonymous call feature for account X.
Page 179 Configuring Basic Features (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 Call. Press , or the Switch soft key to select the desired line from the Account ID field.
Page 180 Administrator’s Guide for SIP-T2xP IP Phones 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_c account.X.anonymous_reject_oncode account.X.anonymous_reject_offcode Configure anonymous call rejection. Navigate to: Web User Interface http://<phoneIPAddress>/servlet?p=acc...
Page 181 Configuring Basic Features Parameters Permitted Values Default Description: Configures the anonymous call rejection on code to activate the server-side anonymous call rejection feature for account X. The IP phone will send anonymous call rejection on code to the server when you activate anonymous call rejection feature on the IP phone.
Page 182 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Configures what code sent to the server for account X. 0- off code 1- on code X ranges from 1 to 6 (for SIP-T28P). X ranges from 1 to 3 (for SIP-T26P/T22P).
Page 183 Configuring Basic Features Press , or the Switch soft key to select the desired line from the Account ID field. Press to scroll to the Anon Reject field. Press to select Enabled from the Anon Reject field. Press to scroll to the Send rejection Code field. (Optional.) Press to select the desired value from the Send rejection Code field.
Page 184 Administrator’s Guide for SIP-T2xP IP Phones Assign a DND key. Parameters: memorykey.X.type/ linekey.X.type/ programablekey.X.type Configure the DND mode. Parameter: features.dnd_mode Configure DND in the IP phone mode. <y0000000000xx>.cfg Parameters: features.dnd.enable features.dnd.on_code features.dnd.off_code Specify the return code and the reason of the SIP response message when DND is enabled.
Page 185 Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default account.X.dnd.enable 0 or 1 Description: Enables or disables DND feature for account X when the DND mode is configured as Custom. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will reject incoming calls on account X. X ranges from 1 to 6 (for SIP-T28P).
Page 186 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default when the DND mode is configured as Custom. The IP phone will send the DND off code to the server when you deactivate DND feature for account X on the IP phone.
Page 187 Configuring Basic Features Parameters Permitted Values Default features.dnd.on_code String within 32 characters Blank Description: Configures the DND on code to activate the server-side DND feature when the DND mode is configured as Phone. The IP phone will send the DND on code to the server when you activate DND feature on the IP phone.
Page 188 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Web User Interface: Features->General Information->Return Code When DND Phone User Interface: None DND Key For more information on how to configure the DSS Key, refer to Appendix D: Configuring DSS Key on page 513.
Page 189 Configuring Basic Features Parameter Permitted Values Default When X=1, the default value is 28 ( History When X=2, the default value is 61 ( Directory When X=3, the default value is 5 ( When X=4, the default value is 30 ( Menu When X=5, the default value is 28 ( History...
Page 190 Administrator’s Guide for SIP-T2xP IP Phones Parameter Permitted Values Default When X=14, the default value is 2 ( Forward Web User Interface: DSSKey->Memory Key/Line Key/Programable Key->Type Phone User Interface: Menu->Features->DSS Keys->Memory Keys (or Line Keys)->DSS Key X (or Line Key X)->Type...
Page 191 Configuring Basic Features 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 radio box in the DND Status field. 3) (Optional.) Enter the DND on code in the DND On Code field.
Page 192 Administrator’s Guide for SIP-T2xP IP Phones To specify the return code and the reason when DND is enabled via web user interface: Click on Features->General Information. Select the desired type from the pull-down list of Return Code When DND. Click Confirm to accept the change.
Page 193 Configuring Basic Features Busy tone is audible to the other party, indicating that the call connection has been broken when one party releases a call. Busy tone delay can define a period of time during which the busy tone is audible. Procedure Busy tone delay can be configured using the configuration files or locally.
Page 194 Administrator’s Guide for SIP-T2xP IP Phones 2． Select the desired value from the pull-down list of Busy Tone Delay (Seconds). 3． Click Confirm to accept the change. Return code when refuse defines the return code and reason of the SIP response message for the refused call.
Page 195 Configuring Basic Features Details of the Configuration Parameter: Parameter Permitted Values Default features.normal_refuse_code 404, 480 or 486 Description: Configures a return code and reason of SIP response messages when the IP phone rejects an incoming call. A specific reason is displayed on the caller’s phone LCD screen.
Page 196 Administrator’s Guide for SIP-T2xP IP Phones Early media refers to media (e.g., audio and video) played to the caller before a SIP call is actually established. Current implementation supports early media through the 183 message. When the caller receives a 183 message with SDP before the call is established, a media channel is established.
Page 197 Configuring Basic Features Parameter Permitted Values Default Phone User Interface: None To configure 180 ring workaround via web user interface: Click on Features->General Information. 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.
Page 198 Administrator’s Guide for SIP-T2xP IP Phones sip.use_out_bound_in_dialog Specify whether to use outbound proxy in a dialog. Local Web User Interface Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load 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.
Page 199 Configuring Basic Features Select the desired value from the pull-down list of Use Outbound Proxy In Dialog. Click Confirm to accept the change. SIP session timers T1, T2 and T4 are SIP transaction layer timers defined in RFC 3261. Timer T1 is an estimate of the Round Trip Time (RTT) of transactions between a SIP client and SIP server.
Page 200 Administrator’s Guide for SIP-T2xP IP Phones Details of Configuration Parameters: Parameters Permitted Values Default Float from 0.5 account.X.advanced.timer_t1 to10 Description: Configures the SIP session timer T1 (in seconds) for account X. T1 is an estimate of the Round Trip Time (RTT) of transactions between a SIP client and SIP server.
Page 201 Configuring Basic Features Parameters Permitted Values Default Description: Configures the session timer of T4 (in seconds) for account X. T4 represents the maximum duration a message will remain in the network. X ranges from 1 to 6 (for SIP-T28P). X ranges from 1 to 3 (for SIP-T26P/T22P). X ranges from 1 to 2 (for SIP-T20P).
Page 202 Administrator’s Guide for SIP-T2xP 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 Integer from 30 account.X.session_timer.expires 1800 to 7200 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-T2xP 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 source located anywhere (LAN, Internet) to the held party. Procedure Call hold can be configured using the configuration files or locally. Configure the call hold tone and call hold tone delay. Parameters: features.play_hold_tone.enable features.play_hold_tone.delay <y0000000000xx>.cfg Specify whether RFC 2543 (c=0.0.0.0) outgoing hold Configuration File signaling is used.
Page 206 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Enables or disables the IP phone to play a tone when there is a call on hold. 0-Disabled 1-Enabled Web User Interface: Features->General Information->Play Hold Tone Phone User Interface: None features.play_hold_tone.delay...
Page 207 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 6 (for SIP-T28P). X ranges from 1 to 3 (for SIP-T26P/T22P).
Page 208 Administrator’s Guide for SIP-T2xP IP Phones To configure call hold tone and call hold tone delay via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Play Hold Tone. Enter the desired time in the Play Hold Tone Delay field.
Page 209 Configuring Basic Features Enter the SIP URI (e.g., sip:moh@sip.com) in the Music Server URI field. Click Confirm to accept the change. Call forward allows users to redirect an incoming call to a third party. IP phones redirect an incoming INVITE message by responding with a 302 Moved Temporarily message, which contains a Contact header with a new URI that should be tried.
Page 210 Administrator’s Guide for SIP-T2xP IP Phones IP phones support the redirected call information sent by the SIP server with Diversion header, per draft-levy-sip-diversion-08, or History-info header, per RFC 4244. The Diversion/History-info header is used to inform the IP phone of a call’s history. For...
Page 211 Configuring Basic Features forward.always.off_code forward.busy.enable forward.busy.target forward.busy.on_code forward.busy.off_code forward.no_answer.enable forward.no_answer.target forward.no_answer.timeout forward.no_answer.on_code forward.no_answer.off_code Configure diversion/history-info feature. Parameter: features.fwd_diversion_enable Configure forward international. Parameter: forward.international.enable Configure call forward. Navigate to: http://<phoneIPAddress>/servlet?p =features-forward&q=load Web User Configure diversion/history-info Interface feature. Local Configure forward international. Navigate to: http://<phoneIPAddress>/ servlet?p=features-general&q=load...
Page 212 Administrator’s Guide for SIP-T2xP IP Phones 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 213 Configuring Basic Features 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 214 Administrator’s Guide for SIP-T2xP IP Phones 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 215 Configuring Basic Features 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 216 Administrator’s Guide for SIP-T2xP IP Phones 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 217 Configuring Basic Features Parameters Permitted Values Default Description: Configures ring times (N) to wait before forwarding incoming calls for account X when the call forward mode is configured as Custom. Incoming calls will be forwarded when not answered after N*6 seconds. X ranges from 1 to 6 (for SIP-T28P).
Page 218 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Configures the no answer forward off code to deactivate the server-side no answer forward feature for account X when the call forward mode is configured as Custom. IP phone will send the no answer forward off code to the server when you deactivate no answer forward feature for account X on the IP phone.
Page 219 Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables always forward feature. 0-Disabled 1-Enabled If it is set to 1 (Enabled), incoming calls are forwarded to the destination number immediately. Web User Interface: Features->Forward &DND->Forward->Always Forward->On/Off Phone User Interface: Menu->Features->Call Forward->Always Forward->Always Forward forward.always.target String within 32 characters...
Page 220 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Configures the always forward off code to deactivate the server-side always forward feature. The IP phone will send the always forward off code to the server when you deactivate always forward feature on the IP phone.
Page 221 Configuring Basic Features Parameters Permitted Values Default Description: Configures the busy forward on code to activate the server-side busy forward feature. The IP phone will send the busy forward on code and the pre-configured destination number to the server when you activate busy forward feature on the IP phone.
Page 222 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default forward.no_answer.target String within 32 characters Blank Description: Configures the destination number the IP phone forwards incoming calls to after a period of ring time. Example: forward.no_answer.target = 3603 Web User Interface: Features->Forward &DND->Forward->No Answer Forward->Target...
Page 223 Configuring Basic Features Parameters Permitted Values Default forward.no_answer.off_code String within 32 characters Blank Description: Configures the no answer forward off code to deactivate the server-side no answer forward feature. The IP phone will send the no answer forward off code to the server when you deactivate no answer forward feature on the IP phone.
Page 224 Administrator’s Guide for SIP-T2xP IP Phones To configure call forward via web user interface: Click on Features->Forward & DND. In the Forward block, mark the desired radio box in the Mode field. a) If you mark the Phone radio box: 1) Mark the desired radio box in the Always/Busy/No Answer Forward field.
Page 225 Configuring Basic Features 5) Select the ring time to wait before forwarding from the pull-down list of After Ring Time (0~120s) (only for the no answer forward). Click Confirm to accept the change. To configure Diversion/History-Info feature via web user interface: Click on Features->General Information.
Page 226 Administrator’s Guide for SIP-T2xP IP Phones To configure forward international 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 227 Configuring Basic Features 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. 3) Press , or the Switch soft key to select the ring time to wait before forwarding from the After Ring Time field.
Page 228 Administrator’s Guide for SIP-T2xP IP Phones You can also configure the busy forward for all accounts. After the busy forward was configured for a specific account, do the following: 1) Press to highlight the Busy Forward field. 2) Press the All Lines soft key.
Page 229 Configuring Basic Features Semi-attended transfer is implemented by a REFER method with Replaces in the Refer-To header. Attended Transfer -- Transfer a call with prior consulting. Attended transfer is  implemented by a REFER method with Replaces in the Refer-To header. Normally, call transfer is completed by pressing the transfer key.
Page 230 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Enables or disables the IP phone to complete the blind transfer through on-hook besides pressing the Transfer/Tran soft key or TRAN/TRANSFER key. 0-Disabled 1-Enabled Web User Interface: Features->Transfer->Blind Transfer On Hook...
Page 231 Configuring Basic Features Select the desired values from the pull-down lists of Semi-Attended Transfer, Blind Transfer On Hook and Attended 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). IP phones implement network conference using the REFER method specified in RFC 4579.
Page 232 Administrator’s Guide for SIP-T2xP IP Phones Details of Configuration Parameters: Parameters Permitted Values Default account.X.conf_type 0 or 2 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 233 Configuring Basic Features To configure the network conference via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced. Select Network Conference from the pull-down list of Conference Type. Enter the conference URI in the Conference URI field. Click Confirm to accept the change.
Page 234 Administrator’s Guide for SIP-T2xP IP Phones Procedure Feature key synchronization can be configured using the configuration files or locally. Configure feature key synchronization. Configuration File <y0000000000xx>.cfg Parameters: bw.feature_key_sync Configure network conference. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=features-general&q=load Details of Configuration Parameter:...
Page 235 Configuring Basic Features Select Enabled from the pull-down list of Feature Key Synchronization. Click Confirm to accept the change. For a conference call, all parties drop the call when the conference initiator drops the conference call. For local conference, transfer on conference hang up allows the other two parties to remain connected when the conference initiator drops the conference call.
Page 236 Administrator’s Guide for SIP-T2xP IP Phones Details of the Configuration Parameter: Parameter & Description Permitted Values Default transfer.tran_others_after_conf_enable 0 or 1 Description: Enables or disables the IP phone to transfer the local conference call to the two parties after the conference initiator drops the local conference call.
Page 237 Configuring Basic Features Directed call pickup is used for picking up an incoming call on a specific extension. A user can pick up the incoming call using a directed pickup key or the DPickup soft key (not applicable to SIP-T20P IP phones). This feature depends on support from a SIP server. For many SIP servers, directed call pickup requires a directed pickup code, which can be configured on a phone or a per-line basis.
Page 238 Administrator’s Guide for SIP-T2xP IP Phones http://<phoneIPAddress>/servl et?p=dsskey&q=load&model= Configure directed call pickup feature on a phone basis. Navigate to: http://<phoneIPAddress>/servl et?p=features-callpickup&q=lo Configure directed call pickup code on a per-line basis. Navigate to: http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac Assign a directed call pickup Phone User Interface key.
Page 239 Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables the IP phone to display the DPickup soft key when the IP phone is in the pre-dialing screen. 0-Disabled 1-Enabled Note: It is not applicable to SIP-T20P IP phones. Web User Interface: Features->Call Pickup->Directed Call Pickup Phone User Interface:...
Page 240 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default X ranges from 1 to 10 (for SIP-T28/T26P). For line keys: X ranges from 1 to 6 (for SIP-T28P) X ranges from 1 to 3 (for SIP-T26P/T22P). X ranges from 1 to 2 (for SIP-T20P).
Page 241 Configuring Basic Features Parameters Permitted Values Default When X=3, the default value is 5 (DND). When X=4, the default value is 30 (Menu). When X=5, the default value is 28 (History). When X=6, the default value is 61 (Directory). When X=7, the default value is 31 (Switch Account). When X=8, the default value is 31 (Switch Account).
Page 242 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default For programable keys: X ranges from 1 to 14 (for SIP-T28/T26P) X=1-10, 14 (for SIP-T22P) X=5-12, 14 (for SIP-T20P) Example: memorykey.1.line = 1 Web User Interface: DSSKey->Memory Key/Line Key/Programable Key ->Line Phone User Interface: Menu->Features->DSS Keys->Memory Keys (or Line Keys)->DSS Key X (or Line Key...
Page 243 Configuring Basic Features To configure a directed call pickup key via web user interface: Click on DSSKey->Memory Key (Line Key or Programable Key). SIP-T22P/T20P IP phones only support line keys and programable keys. In the desired DSS key field, select Directed Pickup from the pull-down list of Type. Enter the directed call pickup code followed by the specific extension in the Value field.
Page 244 Administrator’s Guide for SIP-T2xP IP Phones Select the desired account from the pull-down list of Account. Click on Advanced. Enter the directed call pickup code in the Directed Call Pickup Code field. Click Confirm to accept the change. To configure a directed pickup key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
Page 245 Configuring Basic Features Procedure Group call pickup can be configured using the configuration files or locally. Configure the group call pickup feature. Parameters: <MAC>.cfg features.pickup.group_pickup_enable account.X.group_pickup_code features.pickup.group_pickup_code Assign a group call pickup key. Configuration File Parameters: memorykey.X.type/ linekey.X.type/ <y0000000000xx>.cf programablekey.X.type memorykey.X.line/ linekey.X.line/ programablekey.X.line memorykey.X.value/ linekey.X.value/...
Page 246 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Enables or disables the IP phone to display the GPickup soft key when the IP phone is in the pre-dialing screen. 0-Disabled 1-Enabled Note: It is not applicable to SIP-T20P IP phones.
Page 247 Configuring Basic Features Parameters Permitted Values Default Phone User Interface: None Group Call Pickup Key For more information on how to configure the DSS Key, refer to Appendix D: Configuring DSS Key on page 513. Parameters Permitted Values Default memorykey.X.type/ linekey.X.type/ Refer to the programablekey.X.type following content...
Page 248 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default When X=3, the default value is 5 (DND). When X=4, the default value is 30 (Menu). When X=5, the default value is 28 (History). When X=6, the default value is 61 (Directory).
Page 249 Configuring Basic Features Parameters Permitted Values Default DSSKey->Memory Key/ Line Key / Programable Key ->Type Phone User Interface: Menu->Features->DSS Keys->Memory Keys (or Line Keys)->DSS Key X (or Line Key X)->Type Blank for memory memorykey.X.line/ linekey.X.line/ key, 1-6 for lines Integer from 1 to 6 programablekey.X.line 1-6, 1 for programable key...
Page 250 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Configures the group call pickup feature code. For memory keys: X ranges from 1 to 10 ( for SIP-T28/T26P For line keys: X ranges from 1 to 6 (for SIP-T28P) X ranges from 1 to 3 (for SIP-T26P/T22P).
Page 251 Configuring Basic Features Select the desired line from the pull-down list of Line. Click Confirm to accept the change. To configure group call pickup feature on a phone basis via web user interface: Click on Features->Call Pickup. Select the desired value from the pull-down list of Group Call Pickup (not applicable to SIP-T20P IP phones).
Page 252 Administrator’s Guide for SIP-T2xP IP Phones Enter the group call pickup code in the Group Call Pickup Code field. Click Confirm to accept the change. To configure a group pickup key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
Page 253 Configuring Basic Features 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. Configure dialog info call pickup. Configuration File <MAC>.cfg Parameter: account.X.dialoginfo_callpickup...
Page 254 Administrator’s Guide for SIP-T2xP IP Phones Parameter Permitted Values Default X ranges from 1 to 2 (for SIP-T20P). Web User Interface: Account->Advanced->Dialog Info Call Pickup Phone User Interface: None To configure dialog info call pickup via web user interface: Click on Account.
Page 255 Configuring Basic Features Parameters: memorykey.X.type/ linekey.X.type/ programablekey.X.type Assign a recall key. Navigate to: Web User Interface http://<phoneIPAddress>/servlet Local ?p=dsskey&q=load&model=0 Phone User Interface Assign a recall key. Recall Key For more information on how to configure the DSS Key, refer to Appendix D: Configuring DSS Key on page 513.
Page 256 Administrator’s Guide for SIP-T2xP IP Phones Parameter Permitted Values Default For programable keys: For SIP-T28P/T26P IP phones: When X=1, the default value is 28 (History). When X=2, the default value is 61 (Directory). When X=3, the default value is 5 (DND).
Page 257 Configuring Basic Features Parameter Permitted Values Default When X=11, the default value is 0 (NA). When X=12, the default value is 0 (NA). When X=14, the default value is 2 (Forward). Web User Interface: DSSKey->Memory Key/ Line Key / Programable Key ->Type Phone User Interface: Menu->Features->DSS Keys->Memory Keys (or Line Keys)->DSS Key X (or Line Key X)->Type...
Page 258 Administrator’s Guide for SIP-T2xP IP Phones orbit, by pressing a call park key. The current call is placed on hold and can be retrieved on another IP phone. This feature depends on support from a SIP server. Procedure Call park key can be configured using the configuration files or locally.
Page 259 Configuring Basic Features Parameters Permitted Values Default Menu->Features->DSS Keys->Memory Keys (or Line Keys)->DSS Key X (or Line Key X)->Type Blank for memory memorykey.X.line/ linekey.X.line Integer from 1 to 6 key,1-6 for lines 1-6 Description: Configures the desired line to apply the call park key. For the memory key, x ranges from 1 to 10.
Page 260 If the caller already exists in the local directory, the local contact name assigned to the caller should be preferentially displayed and stored in the call log. Calling and For more information on calling line identification presentation, refer to Connected Line Identification Presentation on Yealink IP Phones , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
Page 261 Configuring Basic Features Procedure CLIP can be configured using the configuration files or locally. Configure the presentation of the caller identity. Configuration File <MAC>.cfg Parameter: account.X.cid_source Configure the presentation of the caller identity. Navigate to: Local Web User Interface http://<phoneIPAddress>/servl et?p=account-adv&q=load&ac Details of the Configuration Parameter: Parameter...
Page 262 If the callee already exists in the local directory, the local contact name assigned to the callee should be preferentially displayed. Calling and For more information on connected line identification presentation, refer to Connected Line Identification Presentation on Yealink IP Phones , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
Page 263 Configuring Basic Features Procedure COLP can be configured only using the configuration files. Configure the presentation of the callee’s 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 264 Administrator’s Guide for SIP-T2xP IP Phones DTMF (Dual Tone Multi-frequency), better known as touch-tone, is used for telecommunication signaling over analog telephone lines in the voice-frequency band. DTMF is the signal sent from the IP phone to the network, which is generated when pressing the IP phone’s keypad during a call.
Page 265 Configuring Basic Features same codec as your voice and is audible to conversation partners. SIP INFO DTMF digits are transmitted by the SIP INFO messages when the voice stream is established after a successful SIP 200 OK-ACK message sequence. The SIP INFO message is sent along the signaling path of the call.
Page 266 Administrator’s Guide for SIP-T2xP IP Phones Details of Configuration Parameters: Parameters Permitted Values Default account.X.dtmf.type 0, 1, 2 or 3 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 267 Configuring Basic Features Parameters Permitted Values Default account.X.dtmf.info_type 1, 2 or 3 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 6 (for SIP-T28P). X ranges from 1 to 3 (for SIP-T26P/T22P).
Page 268 Administrator’s Guide for SIP-T2xP IP Phones To configure the method of transmitting DTMF digits via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced. Select the desired value from the pull-down list of DTMF Type.
Page 269 Configuring Basic Features Select the desired value (1-3) from the pull-down list of DTMF Repetition. Click Confirm to accept the change. Suppress DTMF display allows IP phones to suppress the display of DTMF digits. DTMF digits are displayed as “*” on the LCD screen. Suppress DTMF display delay defines whether to display the DTMF digits for a short period of time before displaying as “*”.
Page 270 Administrator’s Guide for SIP-T2xP IP Phones Details of Configuration Parameters: Parameters Permitted Values Default features.dtmf.hide 0 or 1 Description: Enables or disables the IP phone to suppress the display of DTMF digits during an active call. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the DTMF digits are displayed as asterisks.
Page 271 Configuring Basic Features Select the desired value from the pull-down list of Suppress DTMF Display Delay (not applicable to SIP-T20P IP phones). Click Confirm to accept the change. Call transfer is implemented via DTMF on some traditional servers. The IP phone sends specified DTMF digits to the server for transferring calls to third parties.
Page 272 Administrator’s Guide for SIP-T2xP IP Phones Details of Configuration Parameters: Parameters Permitted Values Default features.dtmf.replace_tran 0 or 1 Description: Enables or disables the IP phone to send DTMF sequences for transfer function when pressing the transfer soft key or the TRAN key.
Page 273 Configuring Basic Features Enter the specified DTMF digits in the Tran Send DTMF field. Click Confirm to accept the change. Intercom allows establishing an audio conversation directly. The IP phone can answer intercom calls automatically. This feature depends on support from a SIP server. Intercom is a useful feature in office environments to quickly connect with an operator or secretary.
Page 274: Intercom Key
Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http://<phoneIPAddress>/servlet ?p=dsskey&q=load&model=0 Phone User Interface Assign an intercom key. Intercom Key For more information on how to configure the DSS Key, refer to Appendix D: Configuring DSS Key on page 513. Parameters...
Page 275 Configuring Basic Features Parameters Permitted Values Default X)->Account ID memorykey.X.value/ String within 99 blank linekey.X.value characters Description: Configures the intercom number. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Example: memorykey.1.value = 1008 Web User Interface:...
Page 276 Administrator’s Guide for SIP-T2xP IP Phones Press , or the Switch soft key to select Intercom from the Type field. Select the desired line from the Account ID field. Enter the remote extension number in the Value field. Press the Save soft key to accept the change.
Page 277 Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default features.intercom.allow 0 or 1 Description: Enables or disables the IP phone to automatically answer an incoming intercom call. 0-Disabled 1-Enabled If it is set to 0 (Disabled), the IP phone will reject incoming intercom calls and sends a busy signal to the caller.
Page 278 Administrator’s Guide for SIP-T2xP 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 parameter “ ” is set to 1 (Enabled).
Page 279 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 280 Administrator’s Guide for SIP-T2xP IP Phones...
Page 281 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  Message Waiting Indicator ...
Page 282 Administrator’s Guide for SIP-T2xP 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 283 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 284: Auto Answer
Administrator’s Guide for SIP-T2xP IP Phones Value of N Ring Tone Splash.wav N<1 or N>8 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 285 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. 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 0 or 1...
Page 286 Administrator’s Guide for SIP-T2xP 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:...
Page 287 Configuring Advanced Features Parameters Permitted Values Default version 73 or later. 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 288 Administrator’s Guide for SIP-T2xP 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 289 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 IP phones for the following conditions. Condition Description Dial...
Page 290 Administrator’s Guide for SIP-T2xP IP Phones Procedure Tones can be configured using the configuration files or locally. Configure the tones for the IP phone. Parameters: voice.tone.country voice.tone.dial voice.tone.ring voice.tone.busy voice.tone.congestion Configuration File <y0000000000xx>.cfg voice.tone.callwaiting voice.tone.dialrecall voice.tone.info voice.tone.stutter voice.tone.message (not applicable to SIP-T20P IP phones) voice.tone.autoanswer...
Page 291 Configuring Advanced Features Parameters Permitted Values Default voice.tone.country = Custom Web User Interface: Settings->Tones->Select Country Phone User Interface: None voice.tone.dial String Blank Description: Customizes the dial tone. tonelist = element[,element] [,element]… Where element = [!]Freq1[+Freq2][+Freq3][+Freq4] /Duration Freq: the frequency of the tone (ranges from 200 to 7000 Hz). If it is set to 0Hz, it means the tone is not played.
Page 292 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default None voice.tone.busy String Blank Description: Customizes the tone when the callee is busy. The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”.
Page 293 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 parameter “voice.tone.country” is set to Custom. Web User Interface: Settings->Tones->Dial Recall Phone User Interface:...
Page 294 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Phone User Interface: None voice.tone.message (not applicable to SIP-T20P IP String Blank phones) 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 295 Configuring Advanced Features If you select Custom, you can customize a tone for each condition of the IP phone. Click Confirm to accept the change. Remote phone book is a centrally maintained phone book, stored on the remote server. Users only need the access URL of the remote phone book. The IP phone can establish a connection with the remote server and download the phone book, and then display the remote phone book entries on the phone user interface.
Page 296 Administrator’s Guide for SIP-T2xP IP Phones outgoing/incoming calls. Parameter: features.remote_phonebook.enable Specify how often the IP phone refreshes the local cache of the remote phone book. Parameter: features.remote_phonebook.flash_time Specify whether to refresh the local cache of the remote phone book at a time when accessing the remote phone book.
Page 297 Configuring Advanced Features Parameters Permitted Values Default 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 Note: It is not applicable to SIP-T20P IP phones. Web User Interface: Directory->Remote Phone Book->Display Name Phone User Interface:...
Page 298 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Directory->Remote Phone Book->Incoming/Outgoing Call lookup Phone User Interface: None 0, Integer from 21600 features.remote_phonebook.flash_time 3600 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 299 Configuring Advanced Features Enter the name in the Display Name field. Click Confirm to accept the change To configure Incoming/Outgoing Call lookup and Update Time Interval via web user interface: Click on Directory->Remote Phone Book. Select the desired value from the pull-down list of Incoming/Outgoing Call lookup. Enter the desired time in the Update Time Interval (seconds) field.
Page 300 Administrator’s Guide for SIP-T2xP IP Phones Microsoft Active Directory Application Mode (ADAM)  The biggest plus for LDAP is that users can access the central LDAP directory of the corporation using IP phones. Therefore they do not have to maintain the directory locally.
Page 301 Configuring Advanced Features Procedure LDAP can be configured using the configuration files or locally. Configure LDAP . Parameters: ldap.enable ldap.name_filter ldap.number_filter ldap.tls_mode ldap.host ldap.port ldap.base ldap.user ldap.password ldap.max_hits Configuration File <y0000000000xx>.cfg 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. Parameters: memorykey.X.type/ linekey.X.type/...
Page 302 Administrator’s Guide for SIP-T2xP IP Phones Details of Configuration Parameters: Parameters Permitted Values Default ldap.enable 0 or 1 Description: Enables or disables LDAP feature on the IP phone. 0-Disabled 1-Enabled Note: It is not applicable to SIP-T20P IP phones. Web User Interface: Directory->LDAP->Enable LDAP...
Page 303 Configuring Advanced Features Parameters Permitted Values Default ldap.number_filter = (|(telephoneNumber=%)(Mobile=%)(ipPhone=%)) When the number prefix of the telephoneNumber, Mobile or ipPhone of the contact record matches the search criteria, the record will be displayed on the LCD screen. Note: It is not applicable to SIP-T20P IP phones. Web User Interface: Directory->LDAP->LDAP Number Filter Phone User Interface:...
Page 304 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Integer from 1 to ldap.port 65535 Description: Configures the port of the LDAP server. Example: ldap.port = 389 Note: It is not applicable to SIP-T20P IP phones. Web User Interface: Directory->LDAP->Port...
Page 305 Configuring Advanced Features Parameters Permitted Values Default None String within 99 ldap.password Blank characters Description: Configures the password to login the LDAP server. This parameter can be left blank in case the server allows anonymous to login. Otherwise you will need to provide the password to login the LDAP server. Example: ldap.password = secret Note: It is not applicable to SIP-T20P IP phones.
Page 306 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Configures the name attributes of each record to be returned by the LDAP server. It compresses the search results. You can configure multiple name attributes separated by spaces. Example: ldap.name_attr = cn sn...
Page 307 Configuring Advanced Features Parameters Permitted Values Default Phone User Interface: None ldap.version 2 or 3 Description: Configures the LDAP protocol version supported by the IP phone. Make sure the protocol value corresponds with the version assigned on the LDAP server. Note: It is not applicable to SIP-T20P IP phones.
Page 308 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default ldap.ldap_sort 0 or 1 Description: Enables or disables the IP phone to sort the search results in alphabetical order or numerical order. 0-Disabled 1-Enabled Note: It is not applicable to SIP-T20P IP phones.
Page 309 Configuring Advanced Features Parameters Permitted Values Default For memory keys: The default value is 0. For line keys: The default value is 15. For programable keys: For SIP-T28P/T26P IP phones: When X=1, the default value is 28 (History). When X=2, the default value is 61 (Directory). When X=3, the default value is 5 (DND).
Page 310 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Phone User Interface: Menu->Features->DSS Keys->Memory Keys (or Line Keys)->DSS Key X (or Line Key X)->Type To configure LDAP via web user interface: Click on Directory->LDAP. Enter the values in the corresponding fields.
Page 311 Configuring Advanced Features In the desired DSS key field, select LDAP from the pull-down list of Type. Click Confirm to accept the change. To configure an LDAP key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field.
Page 312 Administrator’s Guide for SIP-T2xP IP Phones Pickup on page 219. Note Visual alert for BLF pickup feature is not applicable to SIP-T20P IP phones. BLF LED Mode BLF LED Mode provides four kinds of definition for the BLF key LED status. The following table lists the LED statuses of the BLF key when BLF LED Mode is set to 0, 1, 2 or 3 respectively.
Page 313 Configuring Advanced Features LED Status Description The monitored user is dialing. The monitored user is talking. Solid green 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 green (1s) phone number.
Page 314 Administrator’s Guide for SIP-T2xP IP Phones LED Status Description The monitored user’s conversation is placed on hold (This LED status requires server support). The monitored user’s conversation is placed on Slow flashing red (1s) hold. The monitored user is idle.
Page 315 Configuring Advanced Features Procedure BLF can be configured using the configuration files or locally. Specify whether to use visual alert and audio alert for BLF pickup. Parameters: features.pickup.blf_visual_enable features.pickup.blf_audio_enable Assign a BLF key. Parameters: memorykey.X.type/ linekey.X.type Configuration File y0000000000xx.cfg memorykey.X.line/ linekey.X.line memorykey.X.value/ linekey.X.value memorykey.X.pickup_value/ linekey.X.pickup_value...
Page 316 Administrator’s Guide for SIP-T2xP IP Phones 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. 0-Disabled 1-Enabled Note: It is not applicable to SIP-T20P IP phones.
Page 317 Configuring Advanced Features BLF Key For more information on how to configure the DSS Key, refer to Appendix D: Configuring DSS Key on page 513. Parameters Permitted Values Default memory key, 0 for memorykey.X.type/ linekey.X.type Integer line key 15 for Description: Configures a DSS key as a BLF key on the IP phone.
Page 318 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Configures the number of the monitored user. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Example: memorykey.1.value = 1008...
Page 319 Configuring Advanced Features (Optional.) Enter the directed call pickup code in the Extension field. Click Confirm to accept the change. To configure visual alert and audio alert for BLF pickup via web user interface: Click on Features->Call Pickup. Select the desired value from the pull-down list of Visual Alert for BLF Pickup. Select the desired value from the pull-down list of Audio Alert for BLF Pickup.
Page 320 Administrator’s Guide for SIP-T2xP IP Phones 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. Click Confirm to accept the change. To configure a BLF key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
Page 321 Configuring Advanced Features Busy Lamp Field (BLF) List allows a list of specific extensions to be monitored for status changes. It enables the monitoring phone to subscribe to a list of users, and receive notifications of the status of monitored users. Different indicators on the monitoring phone show the status of monitored users.
Page 322 Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http://<phoneIPAddress>/servlet?p=ds skey&q=load&model=0 Phone User Assign a BLF List key. Interface Details of Configuration Parameters: Permitted Parameters Default Values String within account.X.blf.blf_list_uri Blank 256 characters Description: Configures the BLF List URI to monitor a list of users for account X.
Page 323 Configuring Advanced Features Permitted Parameters Default Values 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 6 (for SIP-T28P). X ranges from 1 to 3 (for SIP-T26P/T22P). X ranges from 1 to 2 (for SIP-T20P). Example: account.1.blf_list_barge_in_code = *33 Web User Interface:...
Page 324 Administrator’s Guide for SIP-T2xP IP Phones Permitted Parameters Default Values None Phone User Interface: None phone_setting.blf_list_sequence_type 0 or 1 Description: Configures the order of BLF list keys assigned automatically. 0-Line Key->Memory Key->Ext Key 1-Ext Key->Memory Key->Line Key Note: It is only applicable to SIP-T28P, SIP-T26P IP phones.
Page 325 Configuring Advanced Features Parameters Permitted Values Default Menu->Features->DSS Keys->Memory Keys (or Line Keys)->DSS Key X (or Line Key X)->Type memory key, blank for memorykey.X.line/linekey.X.line Integer 1-6 for lines 1-6 Description: Configures the desired line to apply the BLF List key. For the memory key, x ranges from 1 to 10.
Page 326 Administrator’s Guide for SIP-T2xP IP Phones (Optional.) Enter the retrieve call parked code in the BLF List Retrieve call parked Code field. Click Confirm to accept the change. To configure BLF List keys manually via web user interface: Click on DSSKey->Memory Key (Line Key or Programable Key).
Page 327 Configuring Advanced Features Hide Features Access Code feature enables the IP phone to display the feature identifier instead of the dialed feature access code automatically. For example, the dialed call park code will be replaced by the identifier “Call Park” when you park an active call.
Page 328 Administrator’s Guide for SIP-T2xP IP Phones Permitted Parameters Default Values 0-Disabled 1-Enabled Web User Interface: Features->General Information->Hide Feature Access Codes Phone User Interface: None To enable hide feature access codes feature via web user interface: Click on Features->General Information. Select Enabled from the pull-down list of Hide Feature Access Codes.
Page 329 Configuring Advanced Features voice mail number. IP phones do not need to subscribe for message-summary updates. The server automatically sends a message-summary NOTIFY in a new dialog each time the MWI status changes. Procedure Configuration changes can be performed using the configuration files or locally. Configure subscribe for MWI.
Page 330 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default 1-Enabled If it is set to 1 (Enabled), the IP phone will send a SUBSCRIBE message to the server for message-summary updates. X ranges from 1 to 6 (for SIP-T28P).
Page 331 Configuring Advanced Features Parameters Permitted Values Default Note: It works only if the parameters “account.X.subscribe_mwi” is set to 1 (Enabled) and “voice_mail.number.X” is configured. Web User Interface: Account->Advanced->Subscribe MWI To Voice Mail Phone User Interface: None String within 99 voice_mail.number.X Blank characters Description:...
Page 332 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Web User Interface: Account->Advanced->Voice Mail Display Phone User Interface: None To configure subscribe for MWI via web user interface: Click on Account. Select the desired account from the pull-down list of Account.
Page 333 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 334 Administrator’s Guide for SIP-T2xP 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 335 Configuring Advanced Features group name for a paging list key. Parameter: multicast.paging_address.X.label Assign a multicast paging key or a paging list key. Navigate to: http://<phoneIPAddress>/servlet?p =dsskey&q=load&model=0 Specify a multicast codec for the IP phone to send the RTP stream. Navigate to: http://<phoneIPAddress>/servlet?p Web User Interface =features-general&q=load...
Page 336 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default None String within 99 multicast.paging_address.X.ip_address Blank characters Description: Configures the multicast IP address and port number. Example: multicast.paging_address.1.ip_address = 224.5.6.20:10008 multicast.paging_address.2.ip_address = 224.0.0.1:1001 Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
Page 337 Configuring Advanced Features Parameters Permitted Values Default Configures a DSS key as a multicast paging key on the IP phone. The digit 24 stands for the key type Multicast Paging. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Page 338 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Configures a DSS key as a paging list key on the IP phone. The digit 66 stands for the key type Paging List. For the memory key, x ranges from 1 to 10.
Page 339 Configuring Advanced Features In the desired DSS key field, select Paging List from the pull-down list of Type. Click Confirm to accept the change. To configure a codec for multicast paging via web user interface: Click on Features->General Information. Select the desired codec from the pull-down list of Multicast Codec. Click Confirm to accept the change.
Page 340 Administrator’s Guide for SIP-T2xP IP Phones The label will appear on the LCD screen when sending the RTP multicast. Click Confirm to accept the change. To configure a multicast paging key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
Page 341 Configuring Advanced Features The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255. Enter the group name in the Label field. Press the Save soft key to accept the change. Repeat the step 2-6, you can add more paging groups. IP phones can receive an RTP stream from the pre-configured multicast address(es) without involving SIP signaling, and can handle the incoming multicast paging calls differently depending on the configurations of Paging Barge and Paging Priority Active.
Page 342 Administrator’s Guide for SIP-T2xP IP Phones Configure Paging Barge and Paging Priority Active features. Navigate to: http://<phoneIPAddress>/servlet?p=c ontacts-multicastIP&q=load Details of Configuration Parameters: Parameters Permitted Values Default multicast.listen_address.X.ip_address IP address: port Blank (X ranges from 1 to 10) Description: Configures the multicast address and port number that the IP phone listens to.
Page 343 Configuring Advanced Features Parameters Permitted Values Default 1-Enabled If it is set to 1 (Enabled), the IP phone will answer the incoming multicast paging call with a higher priority and ignore that with a lower priority. Web User Interface: Directory->Multicast IP->Paging Priority Active Phone User Interface: None multicast.receive_priority.priority...
Page 344 Administrator’s Guide for SIP-T2xP IP Phones Click Confirm to accept the change. To configure paging barge and paging priority active features via web user interface: Click on Directory->Multicast IP. Select the desired value from the pull-down list of Paging Barge.
Page 345: Url Record
Get /phonerecording.cgi?model=yealink HTTP/1.0\r\n Request Method: GET Request URI: /phonerecording.cgi?model=yealink Request version: HTTP/1.0 Host: 10.1.2.224\r\n User-agent: yealink SIP-T28P 2.72.0.1 00:16:65:11:30:68\r\n If the recording is successfully started, the server will respond with a 200 OK message. Example of a 200 OK message: <YealinkIPPhoneText> <Title>...
Page 346 Administrator’s Guide for SIP-T2xP IP Phones </Text> <YealinkIPPhoneText> If the recording fails for some reasons, for example, the recording box is full, the server will respond with a 200 OK message. Example of a 200 OK message: <YealinkIPPhoneText> <Title> </Title>...
Page 347 Configuring Advanced Features Navigate to: http://<phoneIPAddress>/se rvlet?p=dsskey&q=load&m odel=0 Assign a record key and URL Phone User Interface record key. Record Key For more information on how to configure the DSS Key, refer to Appendix D: Configuring DSS Key on page 513. Parameters Permitted Values Default...
Page 348 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Example: memorykey.1.type = 35 Web User Interface: DSSKey->Memory Key(or Line Key )->Type Phone User Interface: Menu->Features->DSS Keys->Memory Keys (or Line Keys)->DSS Key X (or Line...
Page 349 Configuring Advanced Features In the desired DSS key field, select Record from the pull-down list of Type. Click Confirm to accept the change. To configure a URL record key via web user interface: Click on DSSKey->Memory Key (or Line Key). In the desired DSS key field, select URL Record from the pull-down list of Type.
Page 350 Administrator’s Guide for SIP-T2xP IP Phones To configure a URL record key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key. Press , or the Switch soft key to select URL Record from the Type field.
Page 351 Configuring Advanced Features Hot Desking Key For more information on how to configure the DSS Key, refer to Appendix D: Configuring DSS Key on page 513. Parameters Permitted Values Default memorykey.X.type/ linekey.X.type/ Refer to the following programablekey.X.type content Description: Configures a DSS key as a hot desking key on the IP phone. The digit 34 stands for the key type Hot Desking.
Page 352 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default When X=8, the default value is 31 (Switch Account). When X=9, the default value is 33 (Status). When X=10, the default value is 0 (NA). When X=11, the default value is 0 (NA).
Page 353 Configuring Advanced Features To configure a hot desking key via web user interface: Click on DSSKey->Memory Keys (Line Key). SIP-T22P/T20P IP phones only support line keys. In the desired DSS key field, select Hot Desking from the pull-down list of Type. Click Confirm to accept the change.
Page 354 Administrator’s Guide for SIP-T2xP IP Phones Event Description Off Hook When the IP phone is off hook. On Hook When the IP phone is on hook. Incoming Call When the IP phone receives an incoming call. Outgoing Call When the IP phone places a call.
Page 355 Configuring Advanced Features Event Description When the IP phone completes auto provisioning via Autop Finish power on. An HTTP or HTTPS GET request may contain variable name and variable value, separated by “=”. Each variable value starts with $ in the query part of the URL. The valid URL format is: http(s)://IP address of server/help.xml?variable name=$variable value.
Page 356 Administrator’s Guide for SIP-T2xP IP Phones Variable Value Description The display name of the callee when the IP phone places a call. $display_remote The display name of the caller when the IP phone receives an incoming call. $call_id The call-id of the active call.
Page 357 Configuring Advanced Features action_url.call_terminated action_url.busy_to_idle action_url.idle_to_busy action_url.ip_change action_url.forward_incoming_call action_url.reject_incoming_call action_url.answer_new_incoming_c action_url.transfer_finished action_url.transfer_failed action_url.setup_autop_finish Configure action URL. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet?p =features-actionurl&q=load Details of Configuration Parameters: Parameters Permitted Values Default action_url.setup_completed URL within 511 characters Blank Description: Configures the action URL the IP phone sends after startup. The value format is: http(s)://IP address of server/help.xml? variable name=variable value.
Page 358 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default $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 Phone User Interface: None action_url.registered Blank URL within 511 characters Description: Configures the action URL the IP phone sends after an account is registered.
Page 359 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the action URL the IP phone sends after a register failed. Example: action_url.register_failed = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Register Failed Phone User Interface: None action_url.off_hook URL within 511 characters Blank Description: Configures the action URL the IP phone sends when off hook.
Page 360 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Example: action_url.incoming_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Incoming Call Phone User Interface: None action_url.outgoing_call URL within 511 characters Blank Description: Configures the action URL the IP phone sends when placing a call.
Page 361 Configuring Advanced Features Parameters Permitted Values Default Features->Action URL->Open DND Phone User Interface: None action_url.dnd_off Blank URL within 511 characters Description: Configures the action URL the IP phone sends when DND feature is disabled. Example: action_url.dnd_off = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Close DND Phone User Interface: None...
Page 362 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default None action_url.busy_fwd_on URL within 511 characters Blank Description: Configures the action URL the IP phone sends when busy forward feature is enabled. Example: action_url.busy_fwd_on = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Open Busy Forward...
Page 363 Configuring Advanced Features Parameters Permitted Values Default None action_url.no_answer_fwd_off URL within 511 characters Blank Description: Configures the action URL the IP phone sends when no answer forward feature is disabled. Example: action_url.no_answer_fwd_off = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Close No Answer Forward Phone User Interface: None action_url.transfer_call...
Page 364 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default action_url.attended_transfer_call Blank URL within 511 characters Description: Configures the action URL the IP phone sends when performing an attended/semi-attended transfer. Example: action_url.attended_transfer_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Attended Transfer...
Page 365 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the action URL the IP phone sends when muting a call. Example: action_url.mute = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Mute Phone User Interface: None action_url.unmute URL within 511 characters Blank Description: Configures the action URL the IP phone sends when un-muting a call.
Page 366 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default action_url.call_terminated = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Terminated Phone User Interface: None action_url.busy_to_idle Blank URL within 511 characters Description: Configures the action URL the IP phone sends when changing the state of the IP phone from busy to idle.
Page 367 Configuring Advanced Features Parameters Permitted Values Default action_url.ip_change = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->IP Changed Phone User Interface: None action_url.forward_incoming_call URL within 511 characters Blank Description: Configures the action URL the IP phone sends when forwarding an incoming call. Example: action_url.forward_incoming_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface:...
Page 368 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Web User Interface: Features->Action URL->Answer New-In Call Phone User Interface: None action_url.transfer_finished URL within 511 characters Blank Description: Configures the action URL the IP phone sends when completing a call transfer.
Page 369 Configuring Advanced Features Parameters Permitted Values Default Web User Interface: Features->Action URL->Autop Finish Phone User Interface: None To configure action URL via web user interface: Click on Features->Action URL. 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.
Page 370 Administrator’s Guide for SIP-T2xP IP Phones Variable Value Phone Action ENTER Press the Enter soft key (Except for SIP-T20P). SPEAKER Press the Speakerphone key. F_TRANSFER Transfers a call to another party. VOLUME_UP Increase the volume. VOLUME_DOWN Decrease the volume. MUTE Mute a call.
Page 371 Configuring Advanced Features Variable Value Phone Action Perform a semi-attended/attended transfer to ATrans=xxx xxx. BTrans=xxx Perform a blind transfer to xxx. CALLEND End a call. Get firmware version, registration, DND or forward configuration information. The valid value of “x” is 0 or 1, 0 means you do not need to get configuration information.
Page 372 Administrator’s Guide for SIP-T2xP IP Phones Specify the trusted IP address(es) for sending the action URI to the IP phone. Local Web User Interface Navigate to: http://<phoneIPAddress>/servlet? p=features-remotecontrl&q=load Details of the Configuration Parameter: Parameter Permitted Values Default features.action_uri_limit_ip IP address or any...
Page 373 Configuring Advanced Features Multiple IP addresses are separated by commas. If you enter “any” in this field, the IP phone can receive and handle GET requests from any IP address. If you leave the field blank, the IP phone cannot receive or handle any HTTP GET request. Click Confirm to accept the change.
Page 374 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 375 Working Server: Server 1 is configured with the domain name of the working server. For example, yealink.pbx.com. DNS mechanism is used such that the working server is resolved to multiple SIP servers for failover purpose. The working server is deployed in redundant pairs, designated as primary and secondary servers.
Page 376: Phone Registration
Administrator’s Guide for SIP-T2xP IP Phones secondary server backs up a primary server when the primary server fails and offers the same functionality as the primary server. Fallback Server: Server 2 is configured with the IP address of the fallback server. For example, 192.168.1.15.
Page 377 X ranges from 1 to 6 (for SIP-T28P). X ranges from 1 to 3 (for SIP-T26P/T22P). X ranges from 1 to 2 (for SIP-T20P). Example: account.1.sip_server.1.address = yealink.pbx.com Web User Interface: Account->Register ->SIP Server Y->Server Host Phone User Interface: None account.X.sip_server.Y.port...
Page 378 Administrator’s Guide for SIP-T2xP IP Phones 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 6 (for SIP-T28P). X ranges from 1 to 3 (for SIP-T26P/T22P).
Page 379 Configuring Advanced Features Parameters Permitted Values Default 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 380 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Web User Interface: None Phone User Interface: None account.X.sip_server.Y.failback_timeout 0, 60 to 65535 3600 (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 account.X.sip_server.Y.failback_mode is set to duration.
Page 381 Configuring Advanced Features 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. Configure registration parameters of the selected account in the corresponding fields. Select the desired value from the pull-down list of Transport. Configure parameters of SIP server 1 and SIP server 2 in the corresponding fields.
Page 382 A query. If no port is found through the DNS query, 5060 will be used. The following details the procedures of DNS query for the IP phone to resolve the domain name (e.g., yealink.pbx.com) of working server into the IP address, port and...
Page 383 SRV query next. TCP will be used, targeted to a host determined by an SRV query of “_sip._tcp.yealink.pbx.com”. If the flag of the NAPTR record returned is empty, the IP phone will perform NAPTR query again according to the previous NAPTR query result.
Page 384 The two records also contain a port “5060”, the IP phone uses this port. If the Target is not a numeric IP address, the IP phone performs an A query. So in this case, the IP phone uses “server1.yealink.pbx.com" and “server2.yealink.pbx.com" for the A query.
Page 385 Configuring Advanced Features configured retry count. Procedure SIP Server Domain Name Resolution can be configured using the configuration files or locally. Configure the transport type on the IP phone. Parameters: Configuration File <MAC>.cfg account.X.transport account.X.naptr_build Configure the transport type on the IP phone. Navigate to: Local Web User Interface...
Page 386 Administrator’s Guide for SIP-T2xP IP Phones 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 6 (for SIP-T28P).
Page 387 Configuring Advanced Features The IP phone will always use the results returned from the static DNS cache.  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 388 Administrator’s Guide for SIP-T2xP IP Phones Details of Configuration Parameters: Parameters Permitted Values Default dns_cache_naptr.X.name 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...
Page 389 Domain name Blank (X ranges from 1 to 12) Description: Configures a domain name to be used for the next SRV query in NAPTR record X. Example: dns_cache_naptr.1.replace = _sip._tcp.yealink.pbx.com Web User Interface: None Phone User Interface: None dns_cache_naptr.X.service String within 32...
Page 390 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default SIP+D2S: SIP over SCTP SIPS+D2T: SIPS over TCP Example: dns_cache_naptr.1.service = SIP+D2T Web User Interface: None Phone User Interface: None dns_cache_naptr.X.ttl Integer from 30 to 2147483647 (X ranges from 1 to 12)
Page 391 Domain name Blank (X ranges from 1 to 12) Description: Configures the domain name of the target host for an A query in SRV record X. Example: dns_cache_srv.1.target = server1.yealink.pbx.com Web User Interface: None Phone User Interface: None dns_cache_srv.X.weight Domain name...
Page 392 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Example: dns_cache_srv.1.weight = 1 Web User Interface: None Phone User Interface: None dns_cache_srv.X.ttl Integer from 30 to 2147483647 (X ranges from 1 to 12) Description: Configures the time interval (in seconds) that SRV record X may be cached before the record should be consulted again.
Page 393 Configuring Advanced Features 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 394 Administrator’s Guide for SIP-T2xP IP Phones 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 6 (for SIP-T28P).
Page 395 Seconds until data unit expires. End of LLDPDU Marks end of LLDPDU. Name assigned to the IP phone. System Name The default value is “yealink”. Description of the IP phone. System Description The default value is “yealink”. The supported and enabled capabilities of the IP phone.
Page 396 Administrator’s Guide for SIP-T2xP IP Phones TLV Type TLV Name Description Port VLAN ID, application type, L2 priority Network Policy and DSCP value. Extended Power type, source, priority and value. Power-via-MDI Inventory – Hardware revision of the IP phone. Hardware Revision Inventory –...
Page 397 Configuring Advanced Features 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 398 Administrator’s Guide for SIP-T2xP IP Phones Enter the desired time interval in the Packet Interval (1~3600s) field. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone.
Page 399 DHCP , the IP phone will examine DHCP option for a valid VLAN ID. The predefined option 132 is used to supply the VLAN ID by default. You can customize the DHCP option used to request the VLAN ID. VLAN Feature on Yealink IP Phones For more information on VLAN, refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
Page 400 Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http://<phoneIPAddress>/servlet?p=n etwork-adv&q=load Configure VLAN for the Internet port and PC port. Phone User Interface Configure DHCP VLAN discovery feature. Details of Configuration Parameters: Parameters Permitted Values Default network.vlan.internet_port_enable 0 or 1 Description: Enables or disables VLAN for the Internet (WAN) port.
Page 401 Configuring Advanced Features Parameters Permitted Values Default 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 402 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default 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 403 Configuring Advanced Features Parameters Permitted Values Default Menu->Settings->Advanced Settings (default password: admin) ->Network->VLAN->DHCP VLAN->Option network.vlan.vlan_change.enable 0 or 1 Description: Enables or disables the IP phone to obtain IP address with lower preference of VLAN assignment method or disable VLAN feature when the IP phone cannot obtain IP address with the current VLAN assignment method.
Page 404 Administrator’s Guide for SIP-T2xP IP Phones A dialog box pops up to prompt that the settings will take effect after a reboot. Click OK to reboot the phone. To configure VLAN for PC port via web user interface: Click on Network->Advanced.
Page 405 Configuring Advanced Features The default option is 132. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. Click OK to reboot the phone. To configure VLAN for Internet port (or PC port) via phone user interface: Press Menu->Settings->Advanced Settings (default password: admin) ->Network->VLAN->WAN Port (or PC Port).
Page 406 Administrator’s Guide for SIP-T2xP IP Phones VPN (Virtual Private Network) is a secured private network connection built on top of public telecommunication infrastructure, such as the Internet. It has become more prevalent due to benefits of scalability, reliability, convenience and security. VPN provides remote offices or individual users with secure access to their organization's network.
Page 407 Configuring Advanced Features http://<phoneIPAddress>/servlet?p =network-adv&q=load Phone User Interface Configure VPN feature. Details of Configuration Parameters: Parameters Permitted Values Default network.vpn_enable 0 or 1 Description: 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 408 Administrator’s Guide for SIP-T2xP IP Phones 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.
Page 409 These metrics can be sent between the phones in RTCP-XR packets. These metrics can also be sent in SIP PUBLISH messages to a central voice quality report collector. Two mechanisms for voice quality monitoring are supported by Yealink IP phones: RTCP-XR ...
Page 410 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Phone User Interface: None The VQ-RTCPXR mechanism, complaint with 6035, sends the service quality metric reports contained in SIP PUBLISH messages to the central report collector. Three types of quality reports can be enabled: Session: Generated at the end of a call.
Page 411 Configuring Advanced Features Configure the generation of interval packets. Parameters: phone_setting.vq_rtcpxr.interval_report.enabl phone_setting.vq_rtcpxr_interval_period Configure the generation of alert packets. Parameters: phone_setting.vq_rtcpxr_moslq_threshold_war ning phone_setting.vq_rtcpxr_moslq_threshold_criti phone_setting.vq_rtcpxr_delay_threshold_war ning phone_setting.vq_rtcpxr_delay_threshold_criti Configure the phone to display RTP status showing the voice quality report of the last call on the web user interface.
Page 412 Administrator’s Guide for SIP-T2xP IP Phones phone_setting.vq_rtcpxr_display_remote_cod ec.enable phone_setting.vq_rtcpxr_display_jitter.enable phone_setting.vq_rtcpxr_display_jitter_buffer_ max.enable phone_setting.vq_rtcpxr_display_packets_lost. enable phone_setting.vq_rtcpxr_display_symm_onew ay_delay.enable phone_setting.vq_rtcpxr_display_round_trip_d elay.enable phone_setting.vq_rtcpxr_display_moslq.enabl phone_setting.vq_rtcpxr_display_moscq.enabl Configure the central report collector. Parameters: <MAC>.cfg account.X.vq_rtcpxr.collector_name account.X.vq_rtcpxr.collector_server_host account.X.vq_rtcpxr.collector_server_port Configure VQ-RTCPXR. Configure the phone to display RTP status showing the voice quality report of the last call on the web user interface.
Page 413 Configuring Advanced Features Details of Configuration Parameters: Permitted Parameters Default Values phone_setting.vq_rtcpxr.session_report.enable 0 or 1 Description: Enables or disables the IP phone to send a session quality report to the central report collector at the end of each call. 0-Disabled 1-Enabled Note: It is only applicable to IP phones running firmware version 73 or later.
Page 414 Administrator’s Guide for SIP-T2xP IP Phones Permitted Parameters Default Values Phone User Interface: None phone_setting.vq_rtcpxr_moslq_threshold_warning 15 to 40 Blank Description: Configures the threshold value of listening MOS score (MOS-LQ) multiplied by 10. The threshold value of MOS-LQ causes the phone to send a warning alert quality report to the central report collector.
Page 415 Configuring Advanced Features Permitted Parameters Default Values phone_setting.vq_rtcpxr_delay_threshold_warning 10 to 2000 Blank Description: Configures the threshold value of one way delay (in ms) that causes the phone to send a warning alert quality report to the central report collector. For example, If it is set to 500, when the value of one way delay computed by the phone is less than or equal to 500, the phone will send a waring alert quality report to the central report collector;...
Page 416 Administrator’s Guide for SIP-T2xP IP Phones Permitted Parameters Default Values Description: Enables or disables the voice quality data of the last call to be displayed on web interface at path Status->RTP Status. 0-Disabled 1-Enabled Note: It is only applicable to IP phones running firmware version 73 or later.
Page 417 Configuring Advanced Features Permitted Parameters Default Values Phone User Interface: None phone_setting.vq_rtcpxr_display_stop_time.enable 0 or 1 Description: Enables or disables the phone to display Current Time or Stop Time on the LCD screen. 0-Disabled 1-Enabled Note: It works only if the parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable”...
Page 418 Administrator’s Guide for SIP-T2xP IP Phones Permitted Parameters Default Values Description: Enables or disables the phone to display Remote User on the LCD screen. 0-Disabled 1-Enabled Note: It works only if the parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1” and it is only applicable to IP phones running firmware version 73 or later.
Page 419 Configuring Advanced Features Permitted Parameters Default Values Phone User Interface: None phone_setting.vq_rtcpxr_display_jitter.enable 0 or 1 Description: Enables or disables the phone to display Jitter on the LCD screen. 0-Disabled 1-Enabled Note: It works only if the parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1” and it is only applicable to IP phones running firmware version 73 or later.
Page 420 Administrator’s Guide for SIP-T2xP IP Phones Permitted Parameters Default Values Description: Enables or disables the phone to display Packet lost on the LCD screen. 0-Disabled 1-Enabled Note: It works only if the parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1” and it is only applicable to IP phones running firmware version 73 or later.
Page 421 Configuring Advanced Features 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 parameter “phone_setting.vq_rtcpxr.states_show_on_gui.enable” is set to “1” and it is only applicable to IP phones running firmware version 73 or later.
Page 422 Administrator’s Guide for SIP-T2xP IP Phones Permitted Parameters Default Values Description: Configures the host name of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages for account X. X ranges from 1 to 6 (for SIP-T28P).
Page 423 Configuring Advanced Features Permitted Parameters Default Values Web User Interface: Account->Advanced->VQ RTCP-XR Collector port Phone User Interface: None To configure session report for VQ-RTCPXR via web user interface: Click on Settings->Voice Monitoring. Select the desired value from the pull-down list of VQ RTCP-XR Session Report. Click Confirm to accept the change.
Page 424 Administrator’s Guide for SIP-T2xP IP Phones Enter the desired value in the Period for Interval Report field. Click Confirm to accept the change. 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.
Page 425 Configuring Advanced Features Enter the desired value in the Critical threshold for Delay field. Click Confirm to accept the change. To configure RTP status displayed on the web page via web user interface: Click on Settings->Voice Monitoring. Select the desired value from the pull-down list of Display Report options on Web.
Page 426 Administrator’s Guide for SIP-T2xP IP Phones Click Confirm to accept the change. The RTP status will appear on the web user interface at the path: Status. To configure RTP status displayed on the LCD screen via web user interface: Click on Settings->Voice Monitoring.
Page 427 Configuring Advanced Features column and then click The selected list appears in the Enabled column. Repeat step 2 to add more items to the Enabled column. To remove an item from the Enabled column, select the desired item and then click To adjust the display order of enabled items, select the desired item and then click The LCD screen will display the item(s) in the adjusted order.
Page 428 Administrator’s Guide for SIP-T2xP IP Phones Enter the port of the central report collector in the VQ RTCP-XR Collector port field. Click Confirm to accept the change. Quality of Service (QoS) is the ability to provide different priorities for different packets in the network, allowing the transport of traffic with special requirements.
Page 429: Voice Qos
Configuring Advanced Features simply based on the DiffServ class. The DSCP value ranges from 0 to 63 with each DSCP specifying a particular per-hop behavior (PHB) applicable to a packet. A PHB refers to the packet scheduling, queuing, policing, or shaping behavior of a node on any given packet.
Page 430 Administrator’s Guide for SIP-T2xP IP Phones Procedure QoS can be configured using the configuration files or locally. Configure the DSCPs for voice packets and SIP packets. Configuration File <y0000000000xx>.cfg Parameters: network.qos.rtptos network.qos.signaltos Configure the DSCPs for voice packets and SIP packets.
Page 431 Configuring Advanced Features Parameters Permitted Values Default Web User Interface: Network->Advanced->SIP QoS (0~63) Phone User Interface: None To configure DSCPs for voice packets and SIP packets via web user interface: Click on Network->Advanced. Enter the desired value in the Voice QoS (0~63) field. Enter the desired value in the SIP QoS (0~63) field.
Page 432 Administrator’s Guide for SIP-T2xP IP Phones NAT Traversal NAT traversal is a general term for techniques that establish and maintain IP connections traversing NAT gateways, typically required for client-to-client networking applications, especially for VoIP deployments. STUN is one of the NAT traversal techniques supported by IP phones.
Page 433 Configuring Advanced Features Parameters Permitted Values Default 1-Enabled X ranges from 1 to 6 (for SIP-T28P). X ranges from 1 to 3 (for SIP-T26P/T22P). X ranges from 1 to 2 (for SIP-T20P). Web User Interface: Account->Register->NAT Phone User Interface: None IP address or account.X.nat.stun_server Blank...
Page 434 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default None To configure NAT traversal and STUN server via web user interface: Click on Account->Register. Select the desired account from the pull-down list of Account. Select STUN from the pull-down list of NAT.
Page 435 Configuring Advanced Features authentication. Yealink 802.1X Authentication For more information on 802.1X authentication, refer to available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Procedure 802.1X authentication can be configured using the configuration files or locally. Configure the 802.1X authentication. Parameters: network.802_1x.mode Configuration File <y0000000000xx>.cfg network.802_1x.identity network.802_1x.md5_password...
Page 436 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default take effect. Web User Interface: Network->Advanced->802.1x->802.1x Mode Phone User Interface: Menu->Settings->Advanced Settings (default password: admin) ->Network->802.1x Settings->802.1x Mode network.802_1x.identity String within 32 characters Blank Description: Configures the user name for 802.1x authentication.
Page 437 Configuring Advanced Features Parameters Permitted Values Default Description: Configures the access URL of the CA certificate when the 802.1x authentication method is configured as EAP-TLS, EAP-PEAP/MSCHAPv2, EAP-TTLS/EAP-MSCHAPV2, EAP-PEAP/GTC or EAP-TTLS/EAP-GTC Example : network.802_1x.root_cert_url = http://192.168.1.10/ca.pem Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 438 Administrator’s Guide for SIP-T2xP IP Phones 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 439 Configuring Advanced Features 5) Click Upload to upload the certificates. c) If you select EAP-PEAP/MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system.
Page 440 Administrator’s Guide for SIP-T2xP IP Phones d) If you select EAP-TTLS/EAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system.
Page 441 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 442 Administrator’s Guide for SIP-T2xP IP Phones 4) Click Upload to upload the certificate. 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 443 Configuring Advanced Features messages contain XML-RPC methods defined in the standard for configuration and management of the CPE. TR-069 is intended to support a variety of functionalities to manage a collection of CPEs, including the following primary capabilities: Auto-configuration and dynamic service provisioning ...
Page 444 Administrator’s Guide for SIP-T2xP IP Phones RPC Method Description FactoryReset This method resets the CPE to its factory default state. This method informs the ACS of the completion (either successful or unsuccessful) of a file transfer TransferComplete initiated by an earlier Download or Upload method call.
Page 445 Configuring Advanced Features Parameters Permitted Values Default 0-Disabled 1-Enabled Web User Interface: Settings->TR069->Enable TR069 Phone User Interface: None String within 128 managementserver.username Blank characters Description: Configures the user name for the IP phone to authenticate with the ACS (Auto Configuration Servers). This string is set to the empty string if no authentication is required.
Page 446 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Configures the access URL of the ACS (Auto Configuration Servers). Example: managementserver.url = http://192.168.1.20/acs/ Web User Interface: Settings->TR069->ACS URL Phone User Interface: None String within 128 managementserver.connection_request_username Blank characters Description: Configures the user name for the IP phone to authenticate the incoming connection requests.
Page 447 Configuring Advanced Features Parameters Permitted Values Default to the ACS (Auto Configuration Servers) 0-Disabled 1-Enabled Web User Interface: Settings->TR069->Enable Periodic Inform Phone User Interface: None Integer from 5 to managementserver.periodic_inform_interval 4294967295 Description: Configures the interval (in seconds) for the IP phone to report its configuration to the ACS (Auto Configuration Servers).
Page 448 Administrator’s Guide for SIP-T2xP IP Phones Enter the user name and password authenticated by the IP phone in the Connection Request Username and Connection Request Password fields. Click Confirm to accept the change. IPv6 is the next generation network layer protocol, designed as a replacement for the current IPv4 protocol.
Page 449 Configuring Advanced Features Procedure IPv6 can be configured using the configuration files or locally. Configure the IPv6 address parameters. Parameters: network.ip_address_mode network.ipv6_internet_port.type network.ipv6_internet_port.ip <MAC>.cfg network.ipv6_prefix network.ipv6_internet_port.gateway Configuration File Configure the IPv6 static DNS address. Parameters: network.ipv6_primary_dns network.ipv6_secondary_dns Configure the IPv6 static DNS. <y0000000000xx>.c Parameter: network.ipv6_static_dns_enable...
Page 450 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default 1-IPv6 2-IPv4&IPv6 Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Basic->Internet Port->Mode (IPv4/IPv6) Phone User Interface: Menu->Settings->Advanced Settings (default password: admin) ->Network->WAN Port->IP Mode...
Page 451 Configuring Advanced Features Parameters Permitted Values Default None network.ipv6_internet_port.ip IPv6 address Blank Description: Configures the IPv6 address when the IP address mode is configured as IPv6 or IPv4&IPv6, and the Internet (WAN) port type for IPv6 is configured as Static IP Address.
Page 452 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default network.ipv6_internet_port.gateway = 3036:1:1:c3c7:c11c:5447:23a6:255 Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Basic->IPv6 Config->Static IP Address->Gateway Phone User Interface: Menu->Settings->Advanced Settings (default password: admin)
Page 453 Configuring Advanced Features Parameters Permitted Values Default Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Basic->IPv6 Config->Static IP Address->Secondary DNS Phone User Interface: Menu->Settings->Advanced Settings (default password: admin) ->Network->WAN Port->IPv6->Static IPv6 Client->IPv6 Sec.DNS Or Menu->Settings->Advanced Settings (default password: admin) ->Network->WAN Port->IPv6->DHCP IPv6 Client->Staic DNS(Enabled) ->IPv6...
Page 454 Administrator’s Guide for SIP-T2xP IP Phones (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 a reboot.
Page 455 Configuring Advanced Features Press , or the Switch soft key to select Enabled from the Static DNS field. Enter the desired values in the IPv6 Pri.DNS and IPv6 Sec.DNS fields respectively. Press the Save soft key to accept the change. The IP phone reboots automatically to make settings effective after a period of time.
Page 456 Administrator’s Guide for SIP-T2xP IP Phones...
Page 457 Configuring Audio Features This chapter provides information for making configuration changes for the following audio features: Headset Prior  Dual Headset  Audio Codecs  Acoustic Clarity Technology  Headset prior allows users to use headset preferentially if a headset is physically connected to the IP phone.
Page 458 Administrator’s Guide for SIP-T2xP IP Phones Parameter Permitted Values Default headset mode. The headset mode will not be deactivated until the user presses the HEADSET key again. Web User Interface: Features->General Information->Headset Prior Phone User Interface: None To configure headset prior via web user interface: Click on Features->General Information.
Page 459 Configuring Audio Features Procedure Dual headset can be configured using the configuration files or locally. Configure dual headset. Configuration File <y0000000000xx>.cfg Parameter: features.headset_training Configure dual headset. Navigate to: Local Web User Interface http://<phoneIPAddress>/se rvlet?p=features-general&q =load Details of the Configuration Parameter: Parameter Permitted Values Default...
Page 460 Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of Dual-Headset. Click Confirm to accept the change. CODEC is an abbreviation of COmpress-DECompress, capable of coding or decoding a digital data stream or signal by implementing an algorithm. The object of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining the quality.
Page 461 Configuring Audio Features The following table summarizes the supported audio codecs on IP phones: Codec Algorithm Reference Bit Rate Sample Packetization Rate Time 20ms G722 G.722 RFC 3551 64 Kbps 16 Ksps PCMA G.711 RFC 3551 64 Kbps 8 Ksps 20ms a-law 20ms...
Page 462 Administrator’s Guide for SIP-T2xP IP Phones Codec Configuration Methods Priority RTPmap Web User Interface Configuration Files G723_53 Web User Interface Configuration Files G723_63 Web User Interface Configuration Files G726-16 Web User Interface Configuration Files G726-24 Web User Interface Configuration Files...
Page 463 Configuring Audio Features Navigate to: http://<phoneIPAddress>/servlet? p=account-codec&q=load&acc=0 Details of Configuration Parameters: Parameters Permitted Values Default account.X.codec.Y.enable Refer to the following 0 or 1 content (Y ranges from 1 to 11) Description: Enables or disables the specified codec for account X. 0-Disabled 1-Enabled X ranges from 1 to 6 (for SIP-T28P).
Page 464 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Web User Interface: Account->Codec Phone User Interface: None account.X.codec.Y.payload_type Refer to the Refer to the following following content content (Y ranges from 1 to 11) Description: Configures the codec for account X.
Page 465 Configuring Audio Features Parameters Permitted Values Default Description: Configures the priority of the enabled codec for account X. X ranges from 1 to 6 (for SIP-T28P). X ranges from 1 to 3 (for SIP-T26P/T22P). X ranges from 1 to 2 (for SIP-T20P). For SIP-T20P/T22P/T26P/T28P IP phones: When Y=1, the default value is 2;...
Page 466 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default When Y=4, the default value is 4; When Y=5, the default value is 18; When Y=6, the default value is 9; When Y=7, the default value is 106; When Y=8, the default value is 103;...
Page 467 Configuring Audio Features Repeat the step 4 to add more codecs to the Enable Codecs column. To remove the codec from the Enable Codecs column, select the desired codec and then click To adjust the priority of codecs, select the desired codec and then click Click Confirm to accept the change.
Page 468 Administrator’s Guide for SIP-T2xP IP Phones Acoustic Echo Cancellation (AEC) is used to reduce acoustic echo from a voice call to provide natural full-duplex communication patterns. It also increases the capacity achieved through silence suppression by preventing echo from traveling across a network.
Page 469 Configuring Audio Features Parameter Permitted Values Default Phone User Interface: None To configure AEC via web user interface: Click on Settings->Voice. Select the desired value from the pull-down list of ECHO. 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.
Page 470 Administrator’s Guide for SIP-T2xP IP Phones Procedure VAD can be configured using the configuration files or locally. Configure VAD. Configuration File <y0000000000xx>.cfg Parameter: voice.vad Configure VAD. Navigate to: Local Web User Interface http://<phoneIPAddress>/ servlet?p=settings-voice& q=load Details of the Configuration Parameter:...
Page 471 Configuring Audio Features Click Confirm to accept the change. Comfort Noise Generation (CNG) is used to generate background noise for voice communications during periods of silence in a conversation. It is a part of the silence suppression or VAD handling for VoIP technology. CNG, in conjunction with VAD algorithms, quickly responds when periods of silence occur and inserts artificial noise until voice activity resumes.
Page 472 Administrator’s Guide for SIP-T2xP IP Phones To configure CNG via web user interface: Click on Settings->Voice. 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.
Page 473 Configuring Audio Features jitter buffer and the delay time for jitter buffer. Navigate to: http://<phoneIPAddress>/ servlet?p=settings-voice& q=load Details of Configuration Parameters: Parameters Permitted Values Default voice.jib.adaptive 0 or 1 Description: Configures the type of jitter buffer. 0-Fixed 1-Adaptive Web User Interface: Settings->Voice->JITTER BUFFER->Type Phone User Interface: None...
Page 474 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default None voice.jib.normal Integer from 0 to 400 Description: Configures the normal delay time (in milliseconds) of jitter buffer. Note: It works only if the parameter “voice.jib.adaptive” is set to 0 (Fixed).
Page 475 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 476 Administrator’s Guide for SIP-T2xP IP Phones AES256-SHA  EDH-RSA-DES-CBC3-SHA  EDH-DSS-DES-CBC3-SHA  DES-CBC3-SHA  DHE-RSA-AES128-SHA  DHE-DSS-AES128-SHA  AES128-SHA  IDEA-CBC-SHA  DHE-DSS-RC4-SHA  RC4-SHA  RC4-MD5  EXP1024-DHE-DSS-DES-CBC-SHA  EXP1024-DES-CBC-SHA  EDH-RSA-DES-CBC-SHA  EDH-DSS-DES-CBC-SHA  DES-CBC-SHA  EXP1024-DHE-DSS-RC4-SHA ...
Page 477 A unique server certificate: It is unique to an IP phone (based on the MAC address) and issued by the Yealink Certificate Authority (CA). A generic server certificate: It issued by the Yealink Certificate Authority (CA). Only if no unique certificate exists, the IP phone may send a generic certificate for authentication.
Page 478 Administrator’s Guide for SIP-T2xP IP Phones Common Name Validation feature enables the IP phone to mandatorily validate the common name of the certificate sent by the connecting server. And Security verification rules are compliant with RFC 2818. In TLS feature, we use the terms trusted and server certificate. These are also known as Note CA and device certificates.
Page 479 Configuring Security Features Navigate to: http://<phoneIPAddress>/servlet?p= account-register&q=load&acc=0 Configure trusted certificates feature. Upload the trusted certificates. Navigate to: http://<phoneIPAddress>/servlet?p= trusted-cert&q=load Configure server certificates feature. Upload the server certificates. Navigate to: http://<phoneIPAddress>/servlet?p= server-cert&q=load Details of Configuration Parameters: Parameters Permitted Values Default account.X.transport Integer Description: Configures the type of transport protocol for account X.
Page 480 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default 0-Disabled 1-Enabled the IP phone will authenticate the server certificate based on If it is set to 1 (Enabled), the trusted certificates list. Only when the authentication succeeds, the IP phone will trust the server.
Page 481 Configuring Security Features Parameters Permitted Values Default take effect. Web User Interface: Security->Trusted Certificates->Common Name Validation Phone User Interface: None security.dev_cert 0 or 1 Description: Configures the type of the device certificates for the IP phone to send for TLS authentication 0-Default certificates 1-Custom certificates...
Page 482 Administrator’s Guide for SIP-T2xP IP Phones Parameters Permitted Values Default Description: Configures the access URL of the certificate the IP phone sends for authentication. Example: server_certificates.url = http://192.168.1.20/ca.pem Note: The certificate you want to upload must be in *.pem or *.cer format.
Page 483 Configuring Security Features Select TLS from the pull-down list of Transport. Click Confirm to accept the change. To configure the trusted certificates via web user interface: Click on Security->Trusted Certificates. Select the desired values from the pull-down lists of Only Accept Trusted Certificates, Common Name Validation and CA Certificates.
Page 484 Administrator’s Guide for SIP-T2xP IP Phones Click Browse to select the certificate (*.pem, *.crt, *.cer or *.der) from your local system. Click Upload to upload the certificate. To configure the server certificates via web user interface: Click on Security->Server Certificates.
Page 485 Configuring Security Features Click Browse to select the certificate (*.pem and *.cer) from your local system. Click Upload to upload the certificate. A dialog box pops up to prompt “Success: The Server Certificate has been loaded! Rebooting, please wait…”. Secure Real-Time Transport Protocol (SRTP) encrypts the RTP streams during VoIP phone calls to avoid interception and eavesdropping.
Page 486 Administrator’s Guide for SIP-T2xP 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 487 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 488 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 489: 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 490 Administrator’s Guide for SIP-T2xP IP Phones (Optional.) Click Browse to locate the target directory from your local system in the Target Directory field. The tool uses the file folder “Encrypted” as the target directory by default. (Optional.) Mark the desired radio box in the AES Model field.
Page 491 Configuring Security Features Procedure Decryption method can be configured using the configuration files. Configure the decryption method. Parameter: auto_provision.aes_key_in_file Configure AES keys. Configuration File <y0000000000xx>.cfg Parameters: auto_provision.aes_key_16.com auto_provision.aes_key_16.mac auto_provision.update_file_mode Configure AES keys. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet?p =settings-autop&q=load Details of Configuration Parameters: Parameters Permitted Values...
Page 492 Administrator’s Guide for SIP-T2xP 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 493 Configuring Security Features Parameters Permitted Values Default 1-Enabled Web User Interface: None Phone User Interface: None To configure AES keys via web user interface: Click on Settings->Auto Provision. Enter the values in the Common AES Key and MAC-Oriented AES Key fields. AES keys must be 16 characters and the supported characters contain: 0-9, A-Z, a-z and the following special characters are also supported: # $ % * +, - .
Page 494 Administrator’s Guide for SIP-T2xP IP Phones...
Page 495 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 496 Administrator’s Guide for SIP-T2xP IP Phones Directory Template  Super Search Template  Local Contact File  Remote XML Phone Book  The replace rule template helps with the creation of multiple replace rules. After setup, place the replace rule template to the provisioning server and specify the access URL in the configuration files.
Page 497 Resource Files <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 498 Administrator’s Guide for SIP-T2xP IP Phones <Data DialNowRule="52[0-6]" LineID="1"/> <Data DialNowRule="xxxxxx" LineID=""/> </DialNow> The softkey layout template allows you to customize soft key layout for different call states. The call states include CallFailed, CallIn, Connecting, Dialing, RingBack and Talking. After setup, place the templates to the provisioning server and specify the access URL in the configuration files.
Page 499 Resource Files The following shows an example of the CallFailed template: <CallFailed> <Disable> <Key Type="Empty"/> <Key Type="Switch"/> <Key Type="Cancel"/> </Disable> <Enable> <Key Type="NewCall"/> <Key Type="Empty"/> <Key Type="Empty"/> <Key Type="Empty"/> </Enable> <Default> <Key Type="NewCall"/> <Key Type="Empty"/> <Key Type="Empty"/> <Key Type="Empty"/> </Default> </CallFailed>...
Page 500 Administrator’s Guide for SIP-T2xP IP Phones Procedure Use the following procedures to customize a directory template. Customizing a directory template: Open the template file using an ASCII editor. For each directory list that you want to configure, edit the corresponding string in the file.
Page 501 Resource Files <root_super_search> indicates the start of a template and </root_super_search>  indicates the end of a template. The default display names of the directory lists are Local Directory, History, Remote  Phone Book and LDAP . When specifying the priority of search results, the valid values are 1, 2, 3 and 4. 1 is ...
Page 502 Administrator’s Guide for SIP-T2xP IP Phones You can add contacts one by one on the IP phone directly. You can also add multiple contacts at a time and/or share contacts between IP phones using the local contact template file. After setup, place the template file to the provisioning server and specify the access URL of the template file in the configuration files.
Page 503 Resource Files For each contact that you want to add, add the following string to the file. Each starts on a separate line: <contact display_name="" office_number="" mobile_number="" other_number="" line="" ring="" group_id_name=""/> Where: display_name="" specifies the name of the contact (This value cannot be blank or duplicated).
Page 504 Administrator’s Guide for SIP-T2xP IP Phones When creating a Menu.xml file, learn the following: <YealinkIPPhoneMenu> indicates the start of a remote phone book file and  </YealinkIPPhoneMenu> indicates the end of a remote phone book file. Create the title of a remote phone book between <Title> and </Title>.
Page 505 Resource Files <Name>Department1</Name> <URL>http://10.2.9.1:99/Department.xml</URL> </MenuItem> <MenuItem> <Name>Department2</Name> <URL>http://10.2.9.1:99/Department.xml</URL> </MenuItem> <SoftKeyItem> <Name>#</Name> <URL>http://10.2.9.1:99/Department.xml</URL> </SoftKeyItem> <SoftKeyItem> <Name>*</Name> <URL>http://10.2.9.1:99/Department.xml</URL> </SoftKeyItem> <SoftKeyItem> <Name>1</Name> <URL>http://10.2.9.1:99/Department.xml</URL> </SoftKeyItem> </YealinkIPPhoneMenu> When creating a Department.xml file, learn the following: <YealinkIPPhoneDirectory> indicates the start of a department file and  </YealinkIPPhoneDirectory>...
Page 506 <DirectoryEntry> <Name>Test2</Name> <Telephone>303</Telephone> <Telephone>915980830849</Telephone> </DirectoryEntry> <DirectoryEntry> <Name>Test3</Name> <Telephone>6650</Telephone> <Telephone>915980830849</Telephone> </DirectoryEntry> </YealinkIPPhoneDirectory> Note Yealink supplies a phonebook generation tool to generate a remote XML phone book. Yealink Phonebook Generation Tool User Guide For more information, refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
Page 507 Troubleshooting This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using IP phones. IP phones can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which can help an administrator more easily find the system problem and fix it.
Page 508 Administrator’s Guide for SIP-T2xP 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 509 Troubleshooting Parameters Permitted Values Default None syslog.server IP address or domain name Blank Description: Configures the IP address or domain name of the syslog server when exporting log to the syslog server. Example: syslog.server = 192.168.1.50 Note: It works only if the parameter “syslog.mode” is set to 1 (Server). If you change this parameter, the IP phone will reboot to make the change take effect.
Page 510 Administrator’s Guide for SIP-T2xP IP Phones Select the desired level from the pull-down list of System Log Level. Click Confirm to accept the change. The system log level is set as 6, the informational level. Note Informational level may make some sensitive information accessible (e.g., password-dial number), we recommend that you reset the system log level to 3 after providing the syslog file.
Page 511 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 512 Administrator’s Guide for SIP-T2xP IP Phones Click Export to open file download window, and then save the file to your local system. The following figure shows a portion of a log file- an account registration: You can capture packet in two ways: capturing the packet via web user interface or using the Ethernet software.
Page 513 Troubleshooting Click Stop to stop capturing. 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 514 Administrator’s Guide for SIP-T2xP IP Phones http://<phoneIPAddress> /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 515 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 BIN configuration files via web user interface: Click on Settings->Configuration.
Page 516 Administrator’s Guide for SIP-T2xP IP Phones 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. Click Export to open file download window, and then save the file to your local system.
Page 517 Click Import to import the configuration file. This section describes solutions to common issues that may occur while using the IP phone. Upon encountering a scenario not listed in this section, contact your Yealink reseller for further support. Do one of the following: Ensure that the IP phone is properly plugged into a functional AC outlet.
Page 518 Administrator’s Guide for SIP-T2xP IP Phones ’ Do one of the following: Ensure that the Ethernet cable is plugged into the Internet port on the IP phone and  the Ethernet cable is not loose. Ensure that the Ethernet cable is not damaged.
Page 519 Troubleshooting date manually. If you have poor sound quality/acoustics like intermittent voice, low volume, echo or other noises, the possible reasons could be: Users are seated too far out of recommended microphone range and sound faint,  or are seated too close to sensitive microphones and cause echo. Intermittent voice is mainly caused by packet loss, due to network congestion, and ...
Page 520 Administrator’s Guide for SIP-T2xP IP Phones From: sip:sipsak@<srchost> CSeq: 10 NOTIFY Call-ID: 1234@<srchost> Event: check-sync;reboot=true The IP phone only uses logo file in DOB format, as the DOB format file has a high compression ratio (the size of the uncompressed file compared to that of the compressed file) and can be stored in smaller space.
Page 521 Troubleshooting able to download the CFG files from the provisioning server. PnP depends on support from a SIP server. ’ Do one of the following: Ensure that the configuration is set correctly.  Reboot the phone. Some configurations require a reboot to take effect. ...
Page 522 Administrator’s Guide for SIP-T2xP IP Phones To reset the IP phone via web user interface: Click on Settings->Upgrade. Click Reset to Factory Reset in the Reset to Factory Setting field. The web user interface prompts the message “Do you want to reset to factory?”.
Page 523 Troubleshooting Logo Phone Line Memory Displa Browser Model 3-line (2*15 characters Text Support SIP-T20P and an icon (Non UI) line)
Page 524 Administrator’s Guide for SIP-T2xP IP Phones...
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 provides an authentication mechanism to devices wishing to attach to a LAN or WLAN. ACS (Auto Configuration server)--responsible for auto-configuration of the Central Processing Element (CPE).
Page 526 Administrator’s Guide for SIP-T2xP IP Phones technological innovation and excellence. LAN (Local Area Network)--used to interconnects network devices in a limited area such as a home, school, computer laboratory, or office building. MIB (Management Information Base)--a virtual database used for managing the entities in a communications network.
Page 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-T2xP 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 +10:30 Australia(Lord Howe Islands) +11:00 New Caledonia(Noumea) +11:00 Russia(Srednekolymsk Time) +11:30 Norfolk Island +12:00 New Zealand(Wellington, Auckland) +12:00 Russian(Kamchatka Time) +12:45 New Zealand(Chatham Islands) +13:00 Tonga(Nukualofa) +13:30 Chatham Islands +14:00 Kiribati Yealink IP phones trust the following CAs by default:...
Page 530  Note Yealink endeavors to maintain a built-in list of most common used CA Certificates. Due to memory constraints, we cannot ensure a complete set of certificates. If you are using a certificate from a commercial Certificate Authority not in the list above, you can send a request to your local distributor.
Page 531 Appendix This section provides the DSS key parameters you can configure on IP phones. DSS key consists of memory key, line key and programable key. The following table lists the number of DSS keys you can configure for each phone model: Phone Model Line Key Memory Key...
Page 532 Administrator’s Guide for SIP-T2xP IP Phones Conference  Forward  Transfer  Hold   ReCall   Directed Pickup  Call Park  DTMF  Voice Mail  Speed Dial  Intercom  Line    Group Listening ...
Page 533 Appendix ReCall  SMS (not applicable to SIP-T20P IP phones)  Directed Pickup  Call Park  DTMF  Voice Mail  Speed Dial  Intercom  Line   Group Listening  XML Group (not applicable to SIP-T20P IP ...
Page 534 Administrator’s Guide for SIP-T2xP IP Phones History  Menu  Switch Account  New SMS (not applicable to SIP-T20P IP  phones) Status  LDAP  Prefix (not applicable to SIP-T20P IP phones)  Zero Touch  Local Directory ...
Page 535 Appendix When X=1, the default value is 28 ( History When X=2, the default value is 61 ( Directory When X=3, the default value is 5 ( 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 536 Administrator’s Guide for SIP-T2xP IP Phones 11-DTMF 12-Voice Mail 13-Speed Dial 14-Intercom 15-Line 16-BLF 17-URL 18-Group Listening 22-XML Group 23-Group Pickup 24-Multicast Paging 25-Record 27-XML Browser 28-History 30-Menu 31-Switch Account 32-New SMS (not applicable to SIP-T20P IP phones) 33-Status 34-Hot Desking...
Page 537 Appendix Parameter- linekey.X.line Parameter- programablekey.X.line Configures the desired line to apply the key feature. For memory keys: X ranges from 1 to 10 (for SIP-T28/T26P). For line keys: X ranges from 1 to 6 (for SIP-T28P) X ranges from 1 to 3 (for SIP-T26P/T22P). X ranges from 1 to 2 (for SIP-T20P).
Page 538 Administrator’s Guide for SIP-T2xP IP Phones Phone Lock  Directory  Format Integer For the memory key and programable key, the default value is not applicable. For the line key, when x=1, the default value is 1. Default Value When x=2, the default value is 2.
Page 539 Appendix X=5-12, 14 (for SIP-T20P) Format String Default Value Blank Range String within 99 characters When you assign the Speed Dial to the memory key, this parameter is used to specify Example the number you want to dial out. memorykey.1.value = 1001 Parameter- Configuration File linekey.X.label...
Page 540 Administrator’s Guide for SIP-T2xP IP Phones This parameter is only applicable to BLF feature. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format String Default Value Blank String within 256 characters Range memorykey.1.pickup_value = *88...
Page 541 Configures the second remote phone book. Example memorykey.1.xml_phonebook = 1 This section describes how Yealink IP phones comply with the IETF definition of SIP as described in RFC 3261. This section contains compliance information in the following: RFC and Internet Draft Support ...
Page 542 Administrator’s Guide for SIP-T2xP IP Phones RFC 2782—A DNS RR for specifying the location of services (DNS SRV)  RFC 2806—URLs for Telephone Calls  RFC 2833—RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals  RFC 2915—The Naming Authority Pointer (NAPTR) DNS Resource Record ...
Page 543 Appendix RFC 3665—Session Initiation Protocol (SIP) Basic Call Flow Examples  RFC 3666—SIP Public Switched Telephone Network (PSTN) Call Flows.  RFC 3680—SIP Event Package for Registrations  RFC 3702—Authentication, Authorization, and Accounting Requirements for the SIP  RFC 3711—The Secure Real-time Transport Protocol (SRTP) ...
Page 544 Administrator’s Guide for SIP-T2xP IP Phones RFC 5589—Session Initiation Protocol (SIP) Call Control - Transfer  RFC 5763—Framework for Establishing a Secure Real-time Transport Protocol (SRTP)  RFC 5806—Diversion Indication in SIP  draft-levy-sip-diversion-04.txt—Diversion Indication in SIP  draft-ietf-sip-cc-transfer-05.txt—SIP Call Control - Transfer ...
Page 545 Appendix Method Supported Notes OPTIONS SUBSCRIBE NOTIFY REFER PRACK INFO MESSAGE UPDATE PUBLISH The following SIP request headers are supported: Note In the following table, a “Yes” in the Supported column means the header is sent and properly parsed. Method Supported Notes Accept...
Page 546 Administrator’s Guide for SIP-T2xP IP Phones Method Supported Notes Expires From Max-Forwards Min-SE P-Asserted-Identity P-Preferred-Identity Proxy-Authenticate Proxy-Authorization RAck Record-Route Refer-To Referred-By Remote-Party-ID Replaces Require Route RSeq Session-Expires Subscription-State Supported User-Agent The following SIP responses are supported: Note In the following table, a “Yes” in the Supported column means the header is sent and...
Page 547 Appendix 1xx Response—Information Responses 1xx Response Supported Notes 100 Trying 180 Ringing 181 Call Is Being Forwarded 183 Session Progress 2xx Response—Successful Responses 2xx Response Supported Notes 200 OK 202 Accepted In REFER transfer. 3xx Response—Redirection Responses 3xx Response Supported Notes 300 Multiple Choices 301 Moved Permanently...
Page 548 Administrator’s Guide for SIP-T2xP IP Phones 4xx Response Supported Notes 410 Gone 411 Length Required 413 Request Entity Too Large 414 Request-URI Too Long 415 Unsupported Media Type 416 Unsupported URI Scheme 420 Bad Extension 421 Extension Required 423 Interval Too Brief...
Page 549 Appendix 6xx Response—Global Responses 6xx Response Supported Notes 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable SDP Headers Supported v—Protocol version o—Owner/creator and session identifier a—Media attribute c—Connection information m—Media name and transport address s—Session name t—Active time SIP uses six request methods: INVITE—Indicates a user is being invited to participate in a call session.
Page 550  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 551 Appendix Step Action Description User A sends a SIP INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
Page 552 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.
Page 553 Appendix The call flow scenario is as follows: User A calls User B. User B is busy on the IP phone and unable or unwilling to take another call. The call cannot be set up successfully. User A Proxy Server User B F1.
Page 554 The following figure illustrates the scenario of an unsuccessful call caused by the called user’s no answering. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones.
Page 555 Appendix The call flow scenario is as follows: User A calls User B. User B does not answer the call. User A hangs up. The call cannot be set up successfully. User A Proxy Server User B F1. INVITE B F2.
Page 556 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:...
Page 557 Appendix User B answers the call. User A places 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 558 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description in the Call-ID field. The transaction number within a  single call leg is identified in the CSeq field. The media capability User A is  ready to receive is specified.
Page 559 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 560 Administrator’s Guide for SIP-T2xP 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 561 Appendix Step Action Description 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 562 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description The proxy server sends the SIP ACK to User B. The ACK confirms that the proxy ACK—Proxy Server to User B server has received the 200 OK response. The call session is now active.
Page 563 User A to confirm that User C has received the 200 OK response. The following figure illustrates a successful call between Yealink SIP IP phones in which two parties are in a call and then one of the parties transfers the call to a third party...
Page 564 Administrator’s Guide for SIP-T2xP IP Phones 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 565 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 566 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends an INVITE message to the proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
Page 567 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 568 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 569 Appendix User C answers the call. User A transfers the call to User C. Call is established between User B and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4.
Page 570 Administrator’s Guide for SIP-T2xP 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 571 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 572 Administrator’s Guide for SIP-T2xP 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 573 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 574 Administrator’s Guide for SIP-T2xP 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 575 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 576 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 577 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 578 Administrator’s Guide for SIP-T2xP 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 579 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 580 Administrator’s Guide for SIP-T2xP 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 581 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 582 User B and User C. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: User A calls User B.
Page 583 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 584 Administrator’s Guide for SIP-T2xP 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 585 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 SIP 200 OK response to 200 OK—User B to Proxy the proxy server.
Page 586 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description 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. The proxy server forwards the SIP 200 200OK—Proxy Server to User...
Page 587 Index Numeric Call Completion 180 Ring Workaround Call Forward 802.1X Authentication Call Hold Call Log Call Park Call Recording About This Guide Call Transfer Acoustic Echo Cancellation Call Waiting Action URL Calling Line Identification Presentation Action URI Connected Line Identification Presentation Administrator Password Capturing Packets Always Forward...
Page 588 Administrator’s Guide for SIP-T2xP IP Phones Getting Information from Status Indicators NAT Traversal Getting Started Network Address Translation (NAT) Group Call Pickup Network Conference No Answer Forward Notification Popups H.323 Headset Prior Hide Features Access Code Off Hook Hot Line Dialing...
Page 589 Index SIP Responses SIP Session Description Protocol Usage SIP Session Timer Softkey Layout Specifying the Language to Use SRTP STUN Server Suppress DTMF Display Summary of Changes Table of Contents xiii Time and Date Transfer on Conference Hang Up Transfer via DTMF Transport Layer Security (TLS) Troubleshooting Troubleshooting Methods...

Yealink SIP-T2XP Administrator's Manual

Table of Contents

Product Overview

Getting Started

Configuring Basic Features

Appendix 507

Appendix A: Glossary 507

Appendix B: Time Zones 509

Appendix C: Trusted Certificates 511

Appendix D: SIP 523

Appendix E: SIP Call Flows 531

Quick Links

Summary of Contents for Yealink SIP-T2XP

Page 13: Table Of Contents

Page 19: Product Overview

Page 22: Physical Features Of Ip Phones

Page 29: Getting Started

Page 59: Upgrading Firmware

Page 67: Notification Popups

Page 68: Early Media

Page 77: Configuration Methods

Page 90: Phone User Interface

Page 95: Time Zone

Page 96: Configuration Files

Page 274: Intercom Key

Page 284: Auto Answer

Page 345: Url Record

Page 376: Phone Registration

Page 429: Voice Qos

Page 489: Procedure To Encrypt Configuration Files

Yealink SIP-T2XP Administrator's Manual

Table of Contents

Product Overview

Getting Started

Configuring Basic Features

Appendix 507

Appendix A: Glossary 507

Appendix B: Time Zones 509

Appendix C: Trusted Certificates 511

Appendix D: SIP 523

Appendix E: SIP Call Flows 531

Quick Links

Related Manuals for Yealink SIP-T2XP

Summary of Contents for Yealink SIP-T2XP

Table of Contents