Page 2: Declaration Of Conformity
Copyright Copyright © 2013 YEALINK NETWORK TECHNOLOGY Copyright © 2013 Yealink Network Technology CO., LTD. All rights reserved. No parts of this publication may be reproduced or transmitted in any form or by any means, electronic or mechanical, photocopying, recording, or otherwise, for any purpose, without the express written permission of Yealink Network Technology CO., LTD.
Page 3: Class B Digital Device Or Peripheral
Do not dispose of WEEE as unsorted municipal waste and have to collect such WEEE separately. Customer Feedback We are striving to improve our documentation quality and we appreciate your feedback. Email your opinions and comments to DocsFeedback@yealink.com.
Page 4: Gnu Gpl Information
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:...
Page 5: About This Guide
IP Phones Deployment Guide for BroadWorks Environments, which describes how  to configure the BroadSoft features on the BroadWorks web portal and IP phones. For support or service, please contact your Yealink reseller or go to Yealink Technical Support online http://www.yealink.com/Support.aspx. In This Guide The information detailed in this guide is applicable to the firmware version 71 or higher.
Page 6: Summary Of Changes
Administrator’s Guide for SIP-T2xP IP Phones features on IP phones. Chapter 4, “Configuring Advanced Features” describes how to configure the  advanced features on IP phones. Chapter 5, “Configuring Audio Features” describes how to configure the audio  features on IP phones.
Page 7: Changes For Release 71, Guide Version 71.120
About This Guide Appendix B: Time Zones on page  Changes for Release 71, Guide Version 71.120 Major updates have occurred to the following sections: Configuring DSS Key on page  Changes for Release 71, Guide Version 71.110 The following sections are new for this version: Hot Desking on page ...
Page 8 Administrator’s Guide for SIP-T2xP IP Phones Live Dialpad on page  Auto Answer on page  Call Completion on page  Anonymous Call on page  Anonymous Call Rejection on page  Busy Tone Delay on page  Return Code When Refuse on page ...
Page 9: Changes For Release 70, Guide Version 2.0
About This Guide Changes for Release 70, Guide Version 2.0 The following sections are new for this version: Dialog-Info Call Pickup on page  Web Server Type on page  Tones on page  Hot Desking on page  Action URL on page ...
Page 10 Administrator’s Guide for SIP-T2xP IP Phones...
Page 11: Table Of Contents
VoIP Principle ............................1 SIP Components ............................. 2 SIP IP Phone Models ..........................3 Physical Features of SIP-T2xP IP Phones ..................4 Key Features of SIP-T2xP IP Phones ....................8 Getting Started ................11 Connecting the IP Phones ......................... 11 Initialization Process Overview ......................
Page 12 Administrator’s Guide for SIP-T2xP IP Phones Creating Dial Plan ..........................31 Replace Rule ..........................32 Dial-now ............................33 Area Code ............................. 35 Block Out............................36 Configuring Basic Features ............38 Contrast ..............................39 Backlight ..............................40 User Password ............................42 Administrator Password ........................
Page 13 Table of Contents Group Call Pickup..........................106 Dialog-Info Call Pickup ........................110 Call Return ............................112 Call Park ............................... 113 Web Server Type ..........................114 Calling Line Identification Presentation ..................116 Connected Line Identification Presentation ................. 117 DTMF ..............................118 Suppress DTMF Display ........................
Page 14 Administrator’s Guide for SIP-T2xP IP Phones Headset Prior............................195 Dual Headset ............................196 Audio Codecs ............................ 197 Acoustic Clarity Technology ......................201 Acoustic Echo Cancellation ....................201 Voice Activity Detection ......................202 Comfort Noise Generation ....................... 203 Jitter Buffer ........................... 204 Configuring Security Features ..........
Page 15 Table of Contents Why does the IP phone use DOB format logo file instead of popular BMP, JPG and so on? ............................... 240 How to increase or decrease the volume? ................240 What will happen if I connect both PoE cable and power adapter? Which has the higher priority? ..........................
Page 16 Administrator’s Guide for SIP-T2xP IP Phones Appendix F: Sample Configuration File ..................441 Index ..................447...
Page 17: Product Overview
Product Overview Product Overview This chapter contains the following information about SIP-T2xP IP phones: VoIP Principle  SIP Components  SIP IP Phone Models  VoIP Principle VoIP VoIP (Voice over Internet Protocol) is a technology using the Internet Protocol instead of traditional Public Switch Telephone Network (PSTN) technology for voice communications.
Page 18: Sip Components
Administrator’s Guide for SIP-T2xP IP Phones network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control attributes of an end-to-end call. SIP provides capabilities to: Determine the location of the target endpoint -- SIP supports address resolution, ...
Page 19: Sip Ip Phone Models
SIP-T20P  SIP-T2xP IP phones comply with the SIP standard (RFC 3261), and they can only be used within a network that supports this type of phone. For successfully operating as SIP endpoints in your network, SIP-T2xP IP phones must...
Page 20: Physical Features Of Sip-T2Xp Ip Phones
Routers are configured for VoIP.  VoIP gateways are configured for SIP.  The latest (or compatible) firmware of SIP-T2xP IP phones is available.  A call server is active and configured to receive and send SIP messages.  Physical Features of SIP-T2xP IP Phones This section lists the available physical features of SIP-T2xP IP phones.
Page 21 Product Overview SIP-T26P Physical Features: TI TITAN chipset and TI voice engine 132x64 graphic LCD 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated HD Voice: HD Codec, HD Handset, HD Speaker 45 keys including 13 DSS keys 1xRJ9 (4P4C) handset port 1xRJ9 (4P4C) headset port 2xRJ45 10/100M Ethernet ports 1XRJ12 (6P6C) expansion module port 16 LEDs: 1xpower, 3xline, 1xmessage, 1xheadset, 10xmemory...
Page 22 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/Avaya/Asterisk validated HD Voice: HD Codec, HD Handset, HD Speaker 32 keys including 4 soft keys 1xRJ9 (4P4C) handset port...
Page 23 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/Avaya/Asterisk validated HD Voice: HD Codec, HD Handset, HD Speaker 31 keys including 9 function keys 1xRJ9 (4P4C) handset port 1xRJ9 (4P4C) headset port 2xRJ45 10/100M Ethernet ports...
Page 24: Key Features Of Sip-T2Xp Ip Phones
Administrator’s Guide for SIP-T2xP IP Phones Key Features of SIP-T2xP IP Phones In addition to physical features introduced above, SIP-T2xP 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, 3-way local conference.
Page 25 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 26 Administrator’s Guide for SIP-T2xP IP Phones...
Page 27: Getting Started
Configuring Basic Network Parameters  Creating Dial Plan  Connecting the IP Phones This section introduces how to install SIP-T2xP IP phones with components in packaging contents. Attach the stand Connect the handset and optional headset Connect the network and power Note A headset is not included in packaging contents.
Page 28 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 29 Getting Started Connect the network and power: AC power  Power over Ethernet (PoE)  AC Power 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 30: Initialization Process Overview
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 31: Verifying Startup
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 32: Configuration Methods
Administrator’s Guide for SIP-T2xP IP Phones The message “Initializing, Please Wait” appears on the LCD screen as the IP phone starts up. The main LCD screen displays the following: Time and date  Soft key labels (not supported by the SIP-T20P IP phone) ...
Page 33 CFG file is named after the MAC address of the IP phone. For example, if the MAC address of a SIP-T22P IP phone is 001565113af8, names of these two configuration files must be: y000000000005.cfg and 001565113af8.cfg. The name of the Common CFG file for each SIP-T2xP IP phone model is: SIP-T28P: y000000000000.cfg ...
Page 34: Reading Icons
Administrator’s Guide for SIP-T2xP IP Phones Reading Icons Icons associated with different features may appear on the LCD screen. The following table provides a description for each icon on SIP-T2xP IP phone models. T28P T26P T22P T20P Description Network unavailable...
Page 35: Configuring Basic Network Parameters
Getting Started T28P T26P T22P T20P Description 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 Configuring Basic Network Parameters This section describes how to configure basic network parameters for the IP phone. Note This section mainly introduces IPv4 network parameters.
Page 36 Administrator’s Guide for SIP-T2xP IP Phones and other control information are carried in tagged data items that are stored in the options field of the DHCP message. The data items themselves are also called options. DHCP can be initiated by simply connecting the IP phone with the network. IP phones broadcast DISCOVER messages to request the network information carried in DHCP options, and the DHCP server responds with specific values in corresponding options.
Page 37 Getting Started Procedure DHCP can be configured using the configuration files or locally. Configure DHCP on the IP phone. Configuration File <y0000000000xx>.cfg For more information, refer to DHCP on page 250. Configure DHCP on the IP phone. Navigate to: Web User Interface http://<phoneIPAddress>/servlet Local ?p=network&q=load...
Page 38: Configuring Network Parameters Manually
Administrator’s Guide for SIP-T2xP IP Phones Press the Save soft key to accept the change. The IP phone reboots automatically to make settings effective after a period of time. Configuring Network Parameters Manually If DHCP is disabled or IP phones cannot obtain network parameters from the DHCP server, you need to configure them manually.
Page 39 Getting Started Select desired value from the pull-down list of Mode (IPv4/IPv6). Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after reboot. Click OK to reboot the IP phone. To configure a static IPv4 address via web user interface: Click on Network->Basic.
Page 40: Pppoe
Administrator’s Guide for SIP-T2xP IP Phones To configure the IP address mode via phone user interface: Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN Port. Press to select IPv4, IPv6 or IPv4&IPv6 from the IP Mode field. Press the Save soft key to accept the change.
Page 41 Getting Started Configure PPPoE on the IP Phone User Interface phone.
Page 42: Configuring Transmission Methods Of The Internet Port And Pc Port
Configuring Transmission Methods of the Internet Port and PC Port Two Ethernet ports on the back of the IP phone: Internet port and PC port. Three optional methods of transmission configuration for SIP-T2xP IP phone Internet or PC Ethernet ports: Auto-negotiation ...
Page 43 Getting Started Auto-negotiation Auto-negotiation means that two connected devices choose common transmission parameters (e.g., speed and duplex mode) to transmit voice or data over Ethernet. This process entails devices first sharing transmission capabilities and then selecting the highest performance transmission mode supported by both. You can configure the Internet port and PC port on the IP phone to automatically negotiate during the transmission.
Page 44 Administrator’s Guide for SIP-T2xP IP Phones Procedure The transmission methods of Ethernet ports can be configured using the configuration files or locally. Configure the transmission methods of Ethernet ports. For more information, refer to Configuration File <y0000000000xx>.cfg Internet and PC Ports...
Page 45: Configuring Pc Port Mode
Getting Started Configuring PC Port Mode The PC port on the back of the IP phone is used to connect a PC, which can be configured in one of two modes: Bridge: The IP phone functions as a bridge, and the connected PC appears on the ...
Page 46 Administrator’s Guide for SIP-T2xP IP Phones Mark the desired radio box. If you mark the As Router radio box, you can configure the IP address for the PC port and configure DHCP for the PC attached to the PC port.
Page 47: Creating Dial Plan
Getting Started Creating Dial Plan Regular expression, often called a pattern, is an expression that specifies a set of strings. A regular expression provides a concise and flexible means to “match” (specify and recognize) strings of text, such as particular characters, words, or patterns of characters. Regular expression is used by many text editors, utilities, and programming languages to search and manipulate text based on patterns.
Page 48: Replace Rule
Administrator’s Guide for SIP-T2xP IP Phones "9001$145$2". When you dial out "0012354599" on your phone, the IP phone will replace the number with "90012354599". “$1” means 3 digits in the first parenthesis, that is, “235”. “$2” means 2 digits in the second parenthesis, that is, “99”.
Page 49: Dial-Now
Getting Started Enter the desired line ID in the Account field or leave it blank. If you leave this field blank or enter 0, the replace rule applies to all accounts on the IP phone. Click Add to add the replace rule. Dial-now Dial-now is a string used to match numbers entered by the user.
Page 50 Administrator’s Guide for SIP-T2xP IP Phones Plan on page 258. Create the dial-now rule for the IP phone. Navigate to: http://<phoneIPAddress>/servlet ?p=settings-dialnow&q=load Local Web User Interface Configure the delay time for the dial-now rule. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load To create a dial-now rule via web user interface: Click on Settings->Dial Plan->Dial-now.
Page 51: Area Code
Getting Started 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 Code Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country. When entered numbers match the predefined area code rule, the IP phone will automatically add the area code before the numbers and dial out.
Page 52: Block Out
Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http://<phoneIPAddress>/servlet ?p=settings-areacode&q=load To configure an area code rule via web user interface: Click on Settings->Dial Plan->Area Code. Enter the desired values in the Code, Min Length (1-15) and Max Length (1-15) fields.
Page 53 Getting Started Navigate to: http://<phoneIPAddress>/servlet ?p=settings-blackout&q=load To create a block out rule via web user interface: Click on Settings->Dial Plan->Block Out. Enter the desired value in the BlockOut Number field. Enter the desired line ID in the Account field or leave it blank. If you leave this field blank or enter 0, the block out rule applies to all accounts on the IP phone.
Page 54: Configuring Basic Features
Administrator’s Guide for SIP-T2xP IP Phones Configuring Basic Features This chapter provides information for making configuration changes for the following basic features: Contrast  Backlight  User Password  Administrator Password  Phone Lock  Time and Date  Language ...
Page 55: Contrast
Configuring Basic Features Session Timer  Call Hold  Call Forward  Call Transfer  Network Conference  Transfer on Conference Hang Up  Directed Call Pickup  Group Call Pickup  Dialog-Info Call Pickup  Call Return  Call Park ...
Page 56: Backlight
Administrator’s Guide for SIP-T2xP IP Phones ?p=settings-preference&q=load Configure the contrast of the Phone User Interface LCD screen. To configure contrast via web user interface: Click on Settings->Preference. Select the desired value from the pull-down list of Contrast. Click Confirm to accept the change.
Page 57 Configuring Basic Features Always On: Backlight is turned on permanently.  15, 30, 60 or 120: Backlight is turned off when the IP phone is inactive after a preset  period of time (in seconds), but it is automatically turned on if the status of the IP phone changes or any key is pressed.
Page 58: User Password
Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of Backlight Time (seconds). Click Confirm to accept the change. To configure backlight via phone user interface (only applicable to the SIP-T28P IP phone): Press Menu->Settings->Advanced Settings (password: admin) ->Phone Settings->Backlight.
Page 59: Administrator Password
Configuring Basic Features Change the user password of the IP phone. Local Web User Interface Navigate to: http://<phoneIPAddress>/servlet ?p=security&q=load To change the user password via web user interface: Click on Security->Password. Select user from the pull-down list of User Type. Enter new password in the New Password and Confirm Password fields.
Page 60 Administrator’s Guide for SIP-T2xP IP Phones 264. Change the administrator password. Web User Interface Navigate to: http://<phoneIPAddress>/servlet Local ?p=security&q=load Change the administrator Phone User Interface password. To change the administrator password via web user interface: Click on Security->Password. Select admin from the pull-down list of User Type.
Page 61: Phone Lock
Configuring Basic Features Phone Lock Phone lock is used to lock the IP phone to prevent it from unauthorized use. Once the IP phone is locked, a user must enter the password to unlock it. IP phones offer three types of phone lock: Menu Key, Function Keys and All Keys.
Page 62 Administrator’s Guide for SIP-T2xP IP Phones Configure the type of phone lock. Phone User Interface Assign a keypad lock key. To configure phone lock via web user interface: Click on Features->Phone Lock. Select the desired type from the pull-down list of Keypad Lock Type.
Page 63: Time And Date
Configuring Basic Features In the desired memory key (or line key) field, select Keypad Lock from the pull-down list of Type. Click Confirm to accept the change. To configure the type of phone lock via phone user interface: Press Menu->Settings->Advanced Settings (password: admin) ->Phone Settings->Keypad Lock.
Page 64: Daylight Saving Time
Administrator’s Guide for SIP-T2xP IP Phones the time zone. Daylight Saving Time Daylight Saving Time (DST) is the practice of temporary advancing clocks during the summertime so that evenings have more daylight and mornings have less. Typically clocks are adjusted forward one hour at the start of spring and backward in autumn.
Page 65 Configuring Basic Features Configure the NTP server, time zone and DST. Configure the time and date manually. Web User Interface Configure the time and date formats. Navigate to: http://<phoneIPAddress>/servlet Local ?p=settings-datetime&q=load Configure the NTP server and time zone. Configure the time and date Phone User Interface manually.
Page 66 Administrator’s Guide for SIP-T2xP IP Phones Mark the DST By Week radio box in the Fixed Type field. Select the desired values from the pull-down lists of DST Start Month, DST Start Day of Week, DST Start Day of Week Last in Month, DST Stop Month, DST Stop Day of Week and DST Stop Day of Week Last in Month.
Page 67 Configuring Basic Features Click Confirm to accept the change. To configure the time and data format via web user interface: Click on Settings->Time & Date. Select the desired value from the pull-down list of Time Format. Select the desired value from the pull-down list of Date Format. Click Confirm to accept the change.
Page 68: Language
Administrator’s Guide for SIP-T2xP IP Phones 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 & Date->Manual Settings. Enter the date in the Date field. Enter the time in the Time field.
Page 69: Specifying The Language To Use
Configuring Basic Features The following table lists available languages and associated language packs. Available Language Associated Language Pack English lang+English.txt Deutsch lang-German.txt French lang-French.txt Italian lang-Italian.txt Portuguese lang-Portuguese.txt Polish lang-Polish.txt Spanish lang-Spanish.txt Turkish lang-Turkish.txt Procedure Loading language pack can only be performed using the configuration files. Specify the access URL of the language pack.
Page 70 Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http://<phoneIPAddress>/servlet ?p=settings-preference&q=load Specify the language for the Phone User Interface phone user interface. To specify the language for the web user interface via web user interface: Click on Settings->Preference.
Page 71: Logo Customization
The format of the logo file must be *.dob. Before uploading your custom logo to IP phones, ensure your logo file is correctly formatted. For more information on customizing a logo file, refer to Yealink SIP-T2 Series/T3 Series/VP530 IP Phones Auto Provisioning Guide.
Page 72 Administrator’s Guide for SIP-T2xP IP Phones Procedure The logo shown on the idle screen can be configured using the configuration files or locally. Configure the logo shown on the idle screen. Configuration File <y0000000000xx>.cfg For more information, refer to Logo Customization on page 273.
Page 73: Softkey Layout
Configuring Basic Features To configure a text logo via web user interface (For the SIP-T20P IP phone only): Click on Features->General Information. Select the desired value from the pull-down list of User Logo. Enter the desired text (0~15 characters) in the Text Logo field. Click Confirm to accept the change.
Page 74 Administrator’s Guide for SIP-T2xP IP Phones Call State Default Soft Keys Optional Soft Keys Answer Empty Forward Switch CallIn Silence Reject Empty Empty Empty Switch Connecting Empty Cancel Connecting Transfer Empty Empty Switch SemiAttendTrans Empty Cancel Send Empty History Delete...
Page 75 Configuring Basic Features Call State Default Soft Keys Optional Soft Keys Transfer Empty Resume Switch Hold NewCall Answer Cancel Reject Empty Empty Empty Switch Held Empty Answer Cancel Reject NewCall Transfer Empty Directory PreTrans Delete Switch Cancel Send Empty Empty Empty Switch InConference...
Page 76: Key As Send
Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http://<phoneIPAddress>/servlet ?p=settings-softkey&q=load To configure softkey layout via web user interface: Click on Settings->Softkey Layout. Select the desired value from the pull-down list of Custom Softkey. Select the desired state from the pull-down list of Call States.
Page 77 Configuring Basic Features Procedure Key as send can be configured using the configuration files or locally. Configure the send key. Configure send sound. Configuration File <y0000000000xx>.cfg For more information, refer to as Send on page 275. Configure the send key. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load...
Page 78: Hotline
Administrator’s Guide for SIP-T2xP IP Phones To configure send sound via web user interface: Click on Features->Audio. Select the desired value from the pull-down list of Send Sound. Click Confirm to accept the change. To configure send key via phone user interface: Press Menu->Features->Key as Send.
Page 79 Configuring Basic Features Configure the hotline number. Specify the time (in seconds) the IP phone waits before automatically dial out the hotline Web User Interface number. Navigate to: http://<phoneIPAddress>/servlet Local ?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...
Page 80: Call Log
Administrator’s Guide for SIP-T2xP IP Phones Enter the waiting time (in seconds) in the HotLine Delay field. Press the Save soft key to accept the change. Call Log Call log contains call information such as remote party identification, time and date, and call duration.
Page 81: Missed Call Log
Configuring Basic Features 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 82: Local Directory
Administrator’s Guide for SIP-T2xP IP Phones Configure missed call log feature. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=account-basic&q=load&acc To configure missed call log via web user interface: Click on Account. Select the desired account from the pull-down list of Account.
Page 83 Configuring Basic Features on page 374. Add a group and a contact to the local directory. Navigate to: Web User Interface http://<phoneIPAddress>/servlet Local ?p=contactsbasic&q=load&num =1&group= Add a group and a contact to Phone User Interface the local directory. To add a group to the local directory via web user interface: Click on Directory->Local Directory.
Page 84 Administrator’s Guide for SIP-T2xP IP Phones If Auto is selected, the IP phone will use the first available account when placing calls to the contact from the local directory. Click Add to add the contact. To add a group to the local directory via phone user interface: Press Menu->Directory->Local Directory.
Page 85: Live Dialpad
Configuring Basic Features Live Dialpad 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. Configuration File <y0000000000xx>.cfg For more information, refer to Live...
Page 86 Administrator’s Guide for SIP-T2xP IP Phones enabled. Procedure Call waiting and call waiting tone can be configured using the configuration files or locally. Configure call waiting and call waiting tone. Configuration File <y0000000000xx>.cfg For more information, refer to Call Waiting on page 279.
Page 87 Configuring Basic Features (Optional.) Enter the call waiting off code in the Call Waiting Off Code field. Click Confirm to accept the change. To configure call waiting tone via web user interface: Click on Features->Audio. Select the desired value from the pull-down list of Call Waiting Tone. Click Confirm to accept the change.
Page 88: Auto Redial
Administrator’s Guide for SIP-T2xP IP Phones Waiting field. Press , or the Switch soft key to select the desired value from the Play Tone field. (Optional.) Enter the call waiting on code in the CW On Code field. (Optional.) Enter the call waiting off code in the CW Off Code field.
Page 89: Auto Answer
Configuring Basic Features Enter the desired times in the Auto Redial Times (1~300) field. The default value is 10. Click Confirm to accept the change. To configure auto redial via phone user interface: Press Menu->Features->Auto Redial. Press , or the Switch soft key to select the desired value from the Auto Redial field.
Page 90: Call Completion
Administrator’s Guide for SIP-T2xP IP Phones Configure auto answer. Navigate to: Web User Interface http://<phoneIPAddress>/servlet Local ?p=account-basic&q=load&acc Phone User Interface Configure auto answer. To configure auto answer via web user interface: Click on Account. Select the desired account from the pull-down list of Account.
Page 91 Configuring Basic Features IP phones support call completion using the SUBSCRIBE/NOTIFY method, which is specified in draft-poetzl-sipping-call-completion-00, to subscribe to the busy party and receive notifications oftheir status changes. Procedure Call completion can be configured using the configuration files or locally. Configure call completion.
Page 92: Anonymous Call
Administrator’s Guide for SIP-T2xP IP Phones Completion field. Press the Save soft key to accept the change. Anonymous Call Anonymous call allows the caller to conceal the identity from the callee. The callee’s phone LCD screen prompts an incoming call from anonymity. Anonymous call is configurable on a per-line basis.
Page 93: Anonymous Call Rejection
Configuring Basic Features Phone User Interface Configure anonymous call. To configure anonymous call via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Basic. Select the desired value from the pull-down list of Send Anonymous. Select the desired value from the pull-down list of Anonymous Code.
Page 94 Administrator’s Guide for SIP-T2xP IP Phones LCD screen presents “Anonymity Disallowed”. Anonymous call rejection is configurable on a per-line basis. The anonymous call rejection on code and anonymous call rejection off code configured on IP phones are used to activate/deactivate the server-side anonymous call rejection feature.
Page 95: Do Not Disturb
Configuring Basic Features (Optional.) Enter the anonymous call rejection off code in the Off Code field. Click Confirm to accept the change. To configure anonymous call rejection via phone user interface: Press Menu->Features->Anonymous Call. Press , or the Switch soft key to select the desired line from the Line ID field.
Page 96 Administrator’s Guide for SIP-T2xP IP Phones...
Page 97 Configuring Basic Features Return Message When DND This feature defines the return code and the reason of the SIP response message for the rejected incoming call when DND is enabled on the IP phone. The caller’s phone LCD screen displays the received return code. Procedure DND can be configured using the configuration files or locally.
Page 98 Administrator’s Guide for SIP-T2xP IP Phones To configure a DND key via web user interface: Click on DSSKey->Memory Key (or Line Key). In the desired memory key (or line key) field, select DND from the pull-down list of Type. Click Confirm to accept the change.
Page 99 Configuring Basic Features In the DND block, mark the desired radio box in the Mode field. a) If you mark the Phone radio box: 1) Mark the desired radio box in the DND Status field. 2) (Optional.) Enter the DND on code in the DND On Code field. 3) (Optional.) Enter the DND off code in the DND Off Code field.
Page 100 Administrator’s Guide for SIP-T2xP IP Phones 4) (Optional.) Enter the DND off code in the DND Off Code field. Click Confirm to accept the change. To specify the return code and the reason when DND is enabled via web user interface: Click on Features->General Information.
Page 101: Busy Tone Delay
Configuring Basic Features To configure a DND 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. Press , or the Switch soft key to select DND from the Key Type field.
Page 102: Return Code When Refuse
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 Return code when refuse defines the return code and reason of the SIP response message for the refused call.
Page 103: Early Media
Configuring Basic Features message when refusing a call. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load To specify the return code and the reason when refusing a call via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Return Code When Refuse. Click Confirm to accept the change.
Page 104 Administrator’s Guide for SIP-T2xP IP Phones Procedure 180 ring workaround can be configured using the configuration files or locally. Configure 180 ring workaround. Configuration File <y0000000000xx>.cfg For more information, refer to Ring Workaround on page 289. Configur 180 ring workaround.
Page 105: Use Outbound Proxy In Dialog
Configuring Basic Features Use Outbound Proxy in Dialog An outbound proxy server can receive all initiating request messages and route them to the designated destination. If the IP phone is configured to use an outbound proxy server within a dialog, all SIP request messages from the IP phone will be forced to send to the outbound proxy server.
Page 106: Sip Session Timer
Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of Use Outbound Proxy In Dialog. Click Confirm to accept the change. SIP Session Timer 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 107: Session Timer
Configuring Basic Features 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. Enter the desired value in the SIP Session Timer T1 (0.5~10s) field. The default value is 0.5s. Enter the desired value in the SIP Session Timer T2 (2~40s) field.
Page 108 Administrator’s Guide for SIP-T2xP IP Phones Procedure Session timer can be configured using the configuration files or locally. Configure session timer. Configuration File <MAC>.cfg For more information, refer to Session Timer on page 291. Configure session timer. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet...
Page 109: Call Hold
Configuring Basic Features Call Hold Call hold provides a service of placing an active call on hold. When a call is placed on hold, the IP phone sends an INVITE request with a HOLD SDP to the server. IP phones support two call hold methods, one is RFC 3264, which sets the “a”...
Page 110 Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of RFC 2543 Hold. Click Confirm to accept the change. To configure call hold tone and call hold tone delay via web user interface: Click on Features->General Information.
Page 111: Call Forward
Configuring Basic Features Click Confirm to accept the change. Call Forward Call forward allows users to redirect an incoming call to a third party. IP phones redirect an incoming INVITE message by responding with a 302 Moved Temporarily message, which contains a Contact header with a new URI that should be tried. Three types of call forward: Always Forward -- Forward the incoming call immediately.
Page 112 Administrator’s Guide for SIP-T2xP IP Phones Configure call forward. Navigate to: http://<phoneIPAddress>/serv let?p=features-forward&q=lo Web User Interface Configure forward Local international. Navigate to: http://<phoneIPAddress>/ servlet?p=features-general&q =load Configure call forward. Phone User Interface To configure call forward via web user interface: Click on Features->Forward & DND.
Page 113 Configuring Basic Features 2) Enter the destination number you want to forward in the Target field. 3) Enter the on code and off code in the On Code and Off Code fields. 4) Select the ring time to wait before forwarding from the pull-down list of After Ring Time (only for the no answer forward).
Page 114 Administrator’s Guide for SIP-T2xP IP Phones To configure call forward in phone mode via phone user interface: Press Menu->Features->Call Forward. Press to select the desired forwarding type, and then press the Enter soft key. Depending on your selection: a) If you select Always Forward:...
Page 115 Configuring Basic Features 2) Enter the destination number you want to forward all incoming calls to in the Forward To field. 3) (Optional.) Enter the always forward on code and off code respectively in the On Code and Off Code fields. You can also configure the always forward for all accounts.
Page 116: Call Transfer
Administrator’s Guide for SIP-T2xP IP Phones Press the Save soft key to accept the change. Call Transfer Call transfer enables IP phones to transfer an existing call to another party. IP phones support call transfer using the REFER method specified in RFC 3515 and offer three types of transfer: Blind Transfer -- Transfer a call directly to another party without consulting.
Page 117: Network Conference
Configuring Basic Features Click on Features->Transfer. Select the desired values from the pull-down lists of Semi-Attend Transfer, Blind Transfer On Hook and Semi Attend Transfer On Hook. Click Confirm to accept the change. Network Conference Network conference, also known as centralized conference, provides users with flexibility of call with multiple participants (more than three).
Page 118: Transfer On Conference Hang Up
Administrator’s Guide for SIP-T2xP IP Phones 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. Transfer on Conference Hang Up For local conference, all parties drop the call when the conference initiator drops the conference call.
Page 119: Directed Call Pickup
Configuring Basic Features ?p=features-transfer&q=load To configure Transfer on Conference Hang up via web user interface: Click on Features->Transfer. Select the desired value from the pull-down list of Transfer on Conference Hang up. Click Confirm to accept the change. Directed Call Pickup Directed call pickup is used for picking up an incoming call on a specific extension.
Page 120: Directed Call Pickup On
Administrator’s Guide for SIP-T2xP IP Phones Assign a directed call pickup key. For more information, refer to Directed Call Pickup Key page 382. <y0000000000xx>.cfg Configure directed call pickup feature on a phone basis. For more information, refer to Directed Call Pickup on page 305.
Page 121 Configuring Basic Features Select the desired line from the pull-down list of Line. Click Confirm to accept the change. To configure directed call pickup feature on a phone basis via web user interface: Click on Features->Call Pickup. Select the desired value from the pull-down list of Directed Call Pickup. Enter the directed call pickup code in the Directed Call Pickup Code field.
Page 122: Group Call Pickup
Administrator’s Guide for SIP-T2xP IP Phones 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 123 Configuring Basic Features Procedure Group call pickup can be configured using the configuration files or locally. Configure the group call pickup code on a per-line basis. <MAC>.cfg For more information, refer to Group Call Pickup on page 307. Assign a group call pickup key. Configuration File For more information, refer to Group Call Pickup Key...
Page 124 Administrator’s Guide for SIP-T2xP IP Phones pull-down list of Type. Enter the group call pickup code in the Value field.
Page 125 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. Enter the group call pickup code in the Group Call Pickup Code field.
Page 126: Dialog-Info Call Pickup
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 127 Configuring Basic Features Example of the dialog-info message carried in NOTIFY message: <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="6" state="full" entity="sip:1013@10.2.1.199"> <dialog id="706655206@10.2.8.213" call-id="706655206@10.2.8.213" local-tag="827932784" remote-tag="1887460740" direction="recipient"> <state>early</state> <local> <identity>sip:1013@10.2.1.199</identity> <target uri="sip:1013@10.2.1.199"> </target> </local> <remote> <identity>sip:1011@10.2.1.199</identity> <target uri="sip:1011@10.2.8.213:5063"> </target> </remote> </dialog> </dialog-info> Procedure Dialog-info call pickup can be configured using the configuration files or locally.
Page 128: Call Return
Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of Dialog Info Call Pickup. Click Confirm to accept the change. Call Return Call return, also known as last call return, allows users to place a call back to the last caller.
Page 129: Call Park
Configuring Basic Features In the desired memory key (or line key) field, select Call Return from the pull-down list of Type. Click Confirm to accept the change. To configure a call return key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key.
Page 130: Web Server Type
Administrator’s Guide for SIP-T2xP IP Phones Phone User Interface Assign a call park key. To configure a call park key via web user interface: Click on DSSKey->Memory Key (or Line Key). In the desired memory key (or line key) field, select Call Park from the pull-down list of Type.
Page 131 Configuring Basic Features Web server type can be configured using the configuration files or locally. Configure the web access type, HTTP port and HTTPS port. Configuration File <y0000000000xx>.cfg For more information, refer to Web Server Type on page 308. Configure the web access type, HTTP port and HTTPS port.
Page 132: Calling Line Identification Presentation
Administrator’s Guide for SIP-T2xP IP Phones Click OK to reboot the IP phone. To configure web server type via phone user interface: Press Menu->Settings->Advanced Settings (password: admin) ->Network->Webserver Type. Press , or the Switch soft key to select the desired value from the HTTP Status field.
Page 133: Connected Line Identification Presentation
Configuring Basic Features 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 the Caller ID Source. Click Confirm to accept the change. Connected Line Identification Presentation Connected line identification presentation (COLP) allows IP phones to display the identity of the callee specified for outgoing calls.
Page 134: Dtmf
Administrator’s Guide for SIP-T2xP IP Phones DTMF 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 135 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 136 Administrator’s Guide for SIP-T2xP IP Phones pull-down list of DTMF Info Type. Enter the desired value in the DTMF Payload Type (96~127) field. Click Confirm to accept the change. To configure the number of times to send the end RTP Event packet via web user interface: Click on Features->General Information.
Page 137: Suppress Dtmf Display
Configuring Basic Features Select the desired value (1-3) from the pull-down list of DTMF Repetition. Click Confirm to accept the change. Suppress DTMF Display 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 138: Transfer Via Dtmf
Administrator’s Guide for SIP-T2xP IP Phones To configure suppress DTMF display and suppress DTMF display delay via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Suppress DTMF Display. Select the desired value from the pull-down list of Suppress DTMF Display Delay.
Page 139: Intercom
Configuring Basic Features Click on Features->General Information. Select the desired value from the pull-down list of DTMF Replace Tran. Enter the specified DTMF digits in the Tran Send DTMF field. Click Confirm to accept the change. Intercom Intercom allows establishing an audio conversation directly. The IP phone can answer intercom calls automatically.
Page 140: Incoming Intercom Calls
Administrator’s Guide for SIP-T2xP IP Phones http://<phoneIPAddress>/servlet ?p=dsskey&q=load&model=0 Phone User Interface Assign an intercom key. To configure an intercom key via web user interface: Click on DSSKey->Memory Key (or Line Key). In the desired memory key (or line key) field, select Intercom from the pull-down list of Type.
Page 141 Configuring Basic Features Intercom Mute Intercom Mute allows the IP phone to mute the microphone for incoming intercom calls. Intercom Tone Intercom Tone allows the IP phone to play a warning tone before answering an intercom call. Intercom Barge Intercom Barge allows the IP phone to automatically answer an incoming intercom call while an active call is in progress.
Page 142 Administrator’s Guide for SIP-T2xP IP Phones Select the desired values from the pull-down lists of Accept Intercom, Intercom Mute, Intercom Tone and Intercom Barge. Click Confirm to accept the change. To configure intercom via phone user interface: Press Menu->Features->Intercom. Press , or the Switch soft key to select the desired values from the Accept Intercom, Intercom Mute, Intercom Tone and Intercom Barge fields.
Page 143: Configuring Advanced Features
Configuring Advanced Features Configuring Advanced Features This chapter provides information for making configuration changes for the following advanced features: Distinctive Ring Tones  Tones  Remote Phone Book  LDAP  Busy Lamp Field  Music on Hold  Automatic Call Distribution ...
Page 144 Administrator’s Guide for SIP-T2xP IP Phones phone strips out the URL and keyword parameter and maps them to the appropriate ring tone. Alert-Info headers in the following two formats: Alert-Info: http://localIP/Bellcore-drN Alert-Info: <URL>;info=info text;x-line-id=0 If the Alter-Info header contains the keyword “Bellcore-drN”, the IP phone will play ...
Page 145: Distinctive Ring Tones On
Configuring Advanced Features If the Alert-Info header contains a remote URL, the IP phone will try to download  the WAV ring tone file from the URL and then play the remote ring tone. If it fails to download the file, the IP phone will play the local ring tone associated with info text.
Page 146 Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of Distinctive Ring Tones. Click Confirm to accept the change. To configure the internal ringer text and internal ringer file via web user interface: Click on Settings->Ring.
Page 147: Tones
Configuring Advanced Features Click Confirm to accept the change. Tones When receiving a message, the IP phone will play a warning tone. You can customize tones or select specialized tone sets (vary from country to country) to indicate different conditions of the IP phone. The default tones used on IP phones are the US tone sets. Available tone sets for IP phones: Australia ...
Page 148 Administrator’s Guide for SIP-T2xP IP Phones Chile  Czech ETSI  Configured tones can be heard on IP phones for the following conditions. Condition Description Dial When in the pre-dialing interface Ring Back Ring-back tone Busy When the callee is busy...
Page 149: Remote Phone Book
Configuring Advanced Features If you select Custom, you can customize a tone for each condition of the IP phone. Click Confirm to accept the change. Remote Phone Book Remote phone book is centrally maintained phone book, stored on the remote server. Users only need the access URL of the remote phone book.
Page 150 Administrator’s Guide for SIP-T2xP IP Phones refreshes the local cache of the remote phone book. For more information, refer to Remote Phone Book on page 319. Specify the access URL of the remote phone book. Navigate to: http://<phoneIPAddress>/servl et?p=contacts-remote&q=load Specify whether to query the...
Page 151: Ldap
Configuring Advanced Features Select the desired value from the pull-down list of Search Remote Phonebook Name. Enter the desired time in the Search Flash Time (Seconds) field. Click Confirm to accept the change. LDAP LDAP (Lightweight Directory Access Protocol) is an application protocol for accessing and maintaining information services for the distributed directory over an IP network.
Page 152 Administrator’s Guide for SIP-T2xP IP Phones LDAP Attributes The following table lists the most common attributes used to configure the LDAP lookup on IP phones. Abbreviation Name Description givenName First name LDAP attribute being made up from commonName given name joined to surname.
Page 153 Configuring Advanced Features Select the desired values from the corresponding pull-down list. Click Confirm to accept the change. To configure an LDAP key via web user interface: Click on DSSKey->Memory Key (or Line Key). In the desired memory key (or line key) field, select LDAP from the pull-down list of Type.
Page 154: Busy Lamp Field
Administrator’s Guide for SIP-T2xP IP Phones Busy Lamp Field Busy Lamp Field (BLF) is used to monitor a specific user for status changes on IP phones. For example, you can configure a BLF key on a supervisor’s phone to monitor the phone user status (busy or idle).
Page 155 Configuring Advanced Features LED Status Description Solid green The monitored user is idle. Fast flashing red The monitored user receives an incoming call. Solid red The monitored user is busy. The call is parked against the monitored user’s phone Slow flashing red (1s) number.
Page 156 Administrator’s Guide for SIP-T2xP IP Phones Assign a BLF key. Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&q=load&model=0 Specify whether to use visual alert and audio alert for BLF pickup. Web User Interface Navigate to: Local http://<phoneIPAddress>/servl et?p=features-callpickup&q=lo Configure LED off in idle. Navigate to: http://<phoneIPAddress>/servl...
Page 157 Configuring Advanced Features 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. Click Confirm to accept the change. To configure LED off in idle via web user interface: Click on Features->General Information.
Page 158: Music On Hold
Administrator’s Guide for SIP-T2xP IP Phones Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key. Press , or the Switch soft key to select BLF from the Type field. Press , or the Switch soft key to select the desired line from the Account ID field.
Page 159: Automatic Call Distribution
Configuring Advanced Features Enter the SIP URI (e.g., sip:moh@sip.com) in the Music Server URI field. Click Confirm to accept the change. Automatic Call Distribution Automatic Call Distribution (ACD) enables organizations to manage a large number of phone calls on an individual basis. ACD enables the use of IP phones in a call-center role by automatically distributing incoming calls to available users, or agents.
Page 160 Administrator’s Guide for SIP-T2xP IP Phones For more information, refer to ACD Key on page 390. Configure ACD auto available. For more information, refer to on page 328. Assign an ACD key. Navigate to: http://<phoneIPAddress>/servlet ?p=dsskey&q=load&model=0 Web User Interface Configure ACD auto available.
Page 161: Message Waiting Indicator
Configuring Advanced Features Enter the desired time in ACD Auto Available Timer (0~120s) field. Click Confirm to accept the change. To configure an ACD key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key. Press , or the Switch soft key to select ACD from the Type field.
Page 162 Administrator’s Guide for SIP-T2xP IP Phones Procedure Configuration changes can be performed using the configuration files or locally. Configure subscribe for MWI. Configure subscribe MWI to voice mail. Configuration File <MAC>.cfg For more information, refer to Message Waiting Indicator page 328.
Page 163: Multicast Paging
Configuring Advanced Features Click Confirm to accept the change. To configure subscribe MWI to voice mail via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced. Select the desired value from the pull-down list of Subscribe MWI To Voice Mail. Enter the desired voice number in the Voice Mail field.
Page 164 Administrator’s Guide for SIP-T2xP IP Phones Procedure Configuration changes can be performed using the configuration files or locally. Assign a multicast paging key. For more information, refer to Multicast Paging Key on page 391. Configuration File <y0000000000xx>.cfg Specify a multicast codec for the IP phone to use for multicast RTP.
Page 165: Receiving Rtp Stream
Configuring Advanced Features 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. To configure a multicast paging key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
Page 166 Administrator’s Guide for SIP-T2xP IP Phones in progress. If the parameter is configured as disabled, all incoming multicast paging calls will be automatically ignored. If the parameter is the priority value, the incoming multicast paging calls with higher priority are automatically answered and the ones with lower priority are ignored.
Page 167: Call Recording
Configuring Advanced Features The label will appear on the LCD screen when receiving the RTP multicast. 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. Select the desired value from the pull-down list of Paging Priority Active.
Page 168: Url Record
Administrator’s Guide for SIP-T2xP IP Phones server. IP phones themselves do not have memory to store the recording, what they can do is to trigger the recording and indicate the recording status. Normally, there are 2 main methods to trigger a recording on a certain server. We call them record and URL record.
Page 169 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 2.71.0.140 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 170 Administrator’s Guide for SIP-T2xP IP Phones Procedure Call recording key can be configured using the configuration files or locally. Assign a record key. For more information, refer to Record Key on page 392. Configuration File <y0000000000xx>.cfg Assign a URL record key.
Page 171: Hot Desking
Configuring Advanced Features Enter the URL in the Value field. Click Confirm to accept the change. To configure a record key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field. Press , or the Switch soft key to select Record from the Key Type field.
Page 172: Action Url
Administrator’s Guide for SIP-T2xP IP Phones Hot desking key can be configured using the configuration files or locally. Assign a hot desking key. Configuration File <y0000000000xx>.cfg For more information, refer to Desking Key on page 393. Assign a hot desking key.
Page 173 Configuring Advanced Features or HTTPS GET request. You can specify a URL that triggers a GET request when a specified event occurs. Action URL can only be triggered by the pre-defined events (e.g., log on). The valid URL format is: http(s)://IP address of the server/help.xml?. The following table lists the pre-defined events for action URL.
Page 174 Administrator’s Guide for SIP-T2xP IP Phones Event Description Forward Incoming Call When the IP phone forwards an incoming call. Reject Incoming Call When the IP phone rejects an incoming call. Answer New-In Call When the IP phone answers a new call.
Page 175: Action Url On
Configuring Advanced Features Variable Value Description call. The SIP URI of the callee when the IP phone receives an incoming call. The SIP URI of the callee when the IP phone places a call. $remote The SIP URI of the caller when the IP phone receives an incoming call.
Page 176: Action Uri
Administrator’s Guide for SIP-T2xP IP Phones Enter the action URLs in the corresponding fields. Click Confirm to accept the change. Action URI Opposite to action URL, action URI allows IP phones to interact with web server application by receiving and handling an HTTP or HTTPS GET request. When receiving a GET request, the IP phone will perform the specified action and respond with a 200 OK message.
Page 177 Configuring Advanced Features Variable Value Phone Action Press the line keys (For SIP-T28P, X=6, for SIP-T226/22P, L1-LX X=3, for SIP-T20P, X=2). D1-D10 Press the memory keys (Only for SIP-T28/T26P). F_CONFERENCE Press the CONF key (Except for SIP-T22P). F1-F4 Press the soft keys (Except for SIP-T20P). Press the MESSAGE key.
Page 178 Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http://<phoneIPAddress>/servl et?p=features-remotecontrl&q =load Configure reboot in talking feature. Navigate to: http://<phoneIPAddress>/servl et?p=features-general&q=load To configure the trusted IP address(es) for action URI via web user interface: Click on Features->Remote Control. Enter the IP address or any in the Action URI allow IP List field.
Page 179: Server Redundancy
Configuring Advanced Features Select the desired value from the pull-down list of Reboot In Talking. Click Confirm to accept the change. Server Redundancy Server redundancy is often required in VoIP deployments to ensure continuity of phone service, for events where the server needs to be taken offline for maintenance, the server fails, or the connection between the IP phone and the server fails.
Page 180: Phone Registration
Working Server: Server 1 is configured with the domain name of the working server. For example, yealink.pbx.com. DNS mechanism is used such that the working server is resolved to multiple SIP servers for failover purpose. The working server is deployed in redundant pairs, designated as primary and secondary servers.
Page 181: Server Redundancy On
Configuring Advanced Features unavailable, the secondary server will serve as the working server. Procedure Server redundancy can be configured using the configuration files or locally. Configure the server redundancy on the IP phone. Configuration File <MAC>.cfg For more information, refer to Server Redundancy on page 335.
Page 182: Sip Server Domain Name Resolution
DNS query, 5060 will be used. The following details the procedures of DNS query for the IP phone to resolve the domain name (e.g., yealink.pbx.com) of working server into the IP address, port and transport protocol. NAPTR (Naming Authority Pointer) First, the IP phone sends NAPTR query to get the NAPTR pointer and transport protocol.
Page 183 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 184 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 185: Lldp
Configuring Advanced Features Configure the transport type on the IP phone. Navigate to: Local Web User Interface http://<phoneIPAddress>/servl et?p=account-register&q=load &acc=0 LLDP LLDP (Linker Layer Discovery Protocol) is a vendor-neutral Link Layer protocol, which allows IP phones to receive and/or transmit device-related information to directly connected devices on the network that are also using the protocol, and store the information that is learned about other devices.
Page 186 Administrator’s Guide for SIP-T2xP IP Phones TLV Type TLV Name Description 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 187 Serial number of the IP phone. Number Inventory – Manufacturer name of the IP phone. Manufacturer Name The default value is “yealink”. Inventory – Model Model name of the IP phone. Name Assertion identifier of the IP phone. Asset ID The default value is “asset”.
Page 188: Vlan
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 reboot. Click OK to reboot the IP phone.
Page 189 Configuring Advanced Features VLAN Discovery via DHCP IP phones support VLAN discovery via DHCP. When the VLAN Discovery method is set to DHCP, the IP phone will examine DHCP option for a valid VLAN ID. The predefined option 132 is used to supply the VLAN ID by default. You can customize the DHCP option used to request the VLAN ID.
Page 190 Administrator’s Guide for SIP-T2xP IP Phones Select the desired value (0-7) from the pull-down list of Priority. Click Confirm to accept the change. A dialog box pops up to prompt reboot to make the settings effective. Click OK to reboot the IP phone.
Page 191: Vpn
Configuring Advanced Features To configure DHCP VLAN discovery via web user interface: Click on Network->Advanced. In the VLAN block, select the desired value from the pull-down list of DHCP VLAN Active. Enter the desired option in the Option field. The default option is 132. Click Confirm to accept the change.
Page 192 VPN files are: certificates (ca.crt and client.crt), key (client.key) and the configuration file (vpn.cnf) of the VPN client. For more information on how to package a .tar file, refer to VPN Feature on Yealink IP Phones. Procedure VPN can be configured using the configuration files or locally.
Page 193: Quality Of Service
Configuring Advanced Features Click Browse to locate the tar file from the local system. Click Import to import 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 194: Voice Qos
Administrator’s Guide for SIP-T2xP IP Phones Supporting dedicated bandwidth  Improving loss characteristics  Avoiding and managing network congestion  Shaping network traffic  Setting traffic priorities across the network  The Best-Effort service is the default QoS model in IP networks. It provides no guarantees for data delivering, which means delay, jitter, packet loss and bandwidth allocation are unpredictable.
Page 195 Configuring Advanced Features SIP protocol is used for creating, modifying and terminating two-party or multi-party sessions. To ensure good voice quality, SIP packets emanating from IP phones should be configured with a high transmission priority. DSCPs for voice and SIP packets can be specified respectively. Procedure QoS can be configured using the configuration files or locally.
Page 196: Network Address Translation
Administrator’s Guide for SIP-T2xP IP Phones Network Address Translation Network Address Translation (NAT) is essentially a translation table that maps public IP address and port combinations to private ones. This reduces the need for a large number of public IP addresses. NAT ensures security since each outgoing or incoming request must first go through a translation process.
Page 197: Snmp
SNMP server. IP phones only support the GET request from the SNMP server. The following table lists the basic object identifiers (OIDs) supported by IP phones. Description The textual identification of the contact person for the IP phone, YEALINK-MIB 1.3.6.1.2.1.37459.2.1.1.0 together with the contact information. For example, Sysadmin...
Page 198 Administrator’s Guide for SIP-T2xP IP Phones Description (root@localhost) An administratively-assigned name for the IP phone. If the name is unknown, YEALINK-MIB 1.3.6.1.2.1.37459.2.1.2.0 the value is a zero-length string. For example, IPPHONE The physical location of the IP phone. YEALINK-MIB 1.3.6.1.2.1.37459.2.1.3.0...
Page 199: 802.1X Authentication
Configuring Advanced Features Click on Network->Advanced. In the SNMP block, select the desired value from the pull-down list of Active. Enter the desired port in the Port (1~65535) field. Enter IP address(es) or domain name in the Trusted Address field. Multiple IP addresses are separated by space.
Page 200 Administrator’s Guide for SIP-T2xP IP Phones IP phones support protocols EAP-MD5, EAP-TLS, PEAP-MSCHAPv2 and EAP-TTLS/EAP-MSCHAPv2 for 802.1X authentication. Procedure 802.1X authentication can be configured using the configuration files or locally. Configure the 802.1X authentication. Configuration File <y0000000000xx>.cfg For more information, refer to 802.1X...
Page 201 Configuring Advanced Features In the 802.1x block, select the desired protocol from the pull-down list of 802.1x Mode. a) If you select EAP-MD5: 1) Enter the user name for authentication in the Identity field. 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.
Page 202 Administrator’s Guide for SIP-T2xP IP Phones 5) Click Upload to upload the certificates. c) If you select PEAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field.
Page 203 Configuring Advanced Features 4) Click Upload to upload the certificate. d) If you select EAP-TTLS/EAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system.
Page 204 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 reboot. Click OK to reboot the IP phone.
Page 205: Tr-069 Device Management
Configuring Advanced Features 2) Enter the password for authentication in the MD5 Password field. Click Save to accept the change. The IP phone reboots automatically to make the settings effective after a period of time. TR-069 Device Management TR-069 is a technical specification, defined by the Broadband Forum, which defines a mechanism that encompasses secure auto-configuration of a CPE (Customer-Premises Equipment), as well as incorporates other CPE management functions into a common framework.
Page 206 Administrator’s Guide for SIP-T2xP IP Phones RPC Method Description File types supported by IP phones are: Firmware Image  Configuration File  This method is used to cause the CPE to upload a specified file to the designated location. File types supported by IP phones are:...
Page 207: Ipv6 Support
Configuring Advanced Features Enter the URL of the ACS in the ACS URL field. Select the desired value from the pull-down list of Enable Periodic Inform. Enter the desired time in the Periodic Inform Interval (seconds) field. Enter the user name and password authenticated by the IP phone in the Connection Request Username and Connection Request Password fields.
Page 208 Administrator’s Guide for SIP-T2xP IP Phones Procedure IPv6 can be configured using the configuration files or locally. Configure the IPv6 address assignment method. Configuration File <y0000000000xx>.cfg For more information, refer to IPv6 on page 353. Configure the IPv6 address assignment method.
Page 209 Configuring Advanced Features To configure IPv6 address assignment method via phone user interface: Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN Port. Press to select IPv4&IPv6 or IPv6 from the IP Mode field. Press to highlight IPv6 and press the Enter soft key. Press to select the desired IPv6 address assignment method.
Page 210 Administrator’s Guide for SIP-T2xP IP Phones...
Page 211: Configuring Audio Features
Configuring Audio Features Configuring Audio Features This chapter provides information for making configuration changes for the following audio features: Headset Prior  Dual Headset  Audio Codecs  Acoustic Clarity Technology  Headset Prior Headset prior allows users to use headset preferentially if a headset is physically connected to the IP phone.
Page 212: Dual Headset
Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of Headset Prior. Click Confirm to accept the change. Dual Headset Dual headset allows users to use two headsets on one IP phone. To use this feature, users need to physically connect two headsets to the headset and handset jacks respectively.
Page 213: Audio Codecs
Configuring Audio Features Select the desired value from the pull-down list of Dual-Headset. Click Confirm to accept the change. Audio Codecs 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 214 Administrator’s Guide for SIP-T2xP IP Phones The corresponding attributes of the codec are listed as follows: Codec Configuration Methods Priority RTPmap Configuration Files PCMU Web User Interface Configuration Files PCMA Web User Interface Configuration Files G729 Web User Interface Configuration Files...
Page 215 Configuring Audio Features Procedure Configuration changes can be performed using the configuration files or locally. Configure the codecs to use on a per-line basis. Configure the priority and rtpmap for the enabled codec. Configuration File <MAC>.cfg For more information, refer to Audio Codecs on page 357.
Page 216 Administrator’s Guide for SIP-T2xP IP Phones To adjust the priority of codecs, select the desired codec and then click Click Confirm to accept the change. To configure the ptime on a per-line basis via web user interface: Click on Account.
Page 217: Acoustic Clarity Technology
Configuring Audio Features Acoustic Clarity Technology Acoustic Echo Cancellation Acoustic Echo Cancellation (AEC) is used to remove acoustic echo from a voice communication in order to improve the voice quality. It also increases the capacity achieved through silence suppression by preventing echo from traveling across a network.
Page 218: Voice Activity Detection
Administrator’s Guide for SIP-T2xP IP Phones Voice Activity Detection Voice Activity Detection (VAD) is used in speech processing to detect the presence or absence of human speech. When detecting period of “silence”, VAD replaces that silence efficiently with special packets that indicate silence is occurring. It can facilitate speech processing, and deactivate some processes during non-speech section of an audio session.
Page 219: Comfort Noise Generation
Configuring Audio Features Comfort Noise Generation Comfort Noise Generation (CNG) is used to generate background noise for voice communications during periods of silence in a conversation. It is 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 220: Jitter Buffer
Administrator’s Guide for SIP-T2xP IP Phones Jitter Buffer Jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in even intervals. Jitter is a term indicating variations in packet arrival time, can occur because of network congestion, timing drift or route changes.
Page 221 Configuring Audio Features Enter the fixed delay time for fixed jitter buffer in the Nominal field. Click Confirm to accept the change.
Page 222 Administrator’s Guide for SIP-T2xP IP Phones...
Page 223: Configuring Security Features
Configuring Security Features Configuring Security Features This chapter provides information for making configuration changes for the following security-related features: Transport Layer Security  Secure Real-Time Transport Protocol  Encrypting Configuration Files  Note To use these features correctly, we recommend that IP phones running firmware version 71 or later CANNOT be downgraded to the earlier firmware version.
Page 224 Administrator’s Guide for SIP-T2xP IP Phones to negotiate the security settings for a network connection using the TLS/SSL network protocol. IP phones supports the following cipher suites for TLS 1.0: DHE-RSA-AES256-SHA  DHE-DSS-AES256-SHA  AES256-SHA  EDH-RSA-DES-CBC3-SHA  EDH-DSS-DES-CBC3-SHA ...
Page 225 Configuring Security Features Step1: IP phone sends “Client Hello” message proposing SSL options. Step2: Server responds with “Server Hello” message selecting the SSL options, sends its public key information in “Server Key Exchange” message and concludes its part of the negotiation with “Server Hello Done”...
Page 226 Administrator’s Guide for SIP-T2xP IP Phones Configuration changes can be performed using the configuration files or locally. Configure TLS on a per-line basis. <MAC>.cfg For more information, refer to on page 363. Configure trusted certificates feature. Configure server certificates Configuration File feature.
Page 227 Configuring Security Features To configure TLS on a per-line basis via web user interface: Click on Account->Register. Select the desired account from the pull-down list of Account. 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.
Page 228 Administrator’s Guide for SIP-T2xP IP Phones To upload a trusted certificate via web user interface: Click on Security->Trusted Certificates. Click Browse to select the certificate (*.pem, *.crt, *.cer or *.der) from your local system. Click Upload to upload the certificate.
Page 229: Secure Real-Time Transport Protocol
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 Secure Real-Time Transport Protocol (SRTP) encrypts the RTP streams during VoIP phone calls to avoid interception and eavesdropping.
Page 230 Administrator’s Guide for SIP-T2xP IP Phones answers the call by responding with a 200 OK message which carries the negotiated RTP encryption algorithm. Example of the RTP encryption algorithm carried in the SDP of the 200 OK message: m=audio 11780 RTP/SAVP 0 101...
Page 231: Encrypting Configuration Files
, you can use Yealink-supplied encryption tool platform "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 applicable. For more information, refer to Yealink Configuration Encryption Tool User Guide.
Page 232: Procedure To Encrypt Configuration Files
Administrator’s Guide for SIP-T2xP IP Phones For security, administrator should upload encrypted configuration files, <y0000000000xx_Security>.enc and/or <MAC_Security>.enc files to the root directory of the provisioning server. During auto provisioning, the IP phone requests to download <y0000000000xx>.cfg file first. If the downloaded configuration file is encrypted, the phone will request to download <y0000000000xx_Security>.enc file (if enabled) and...
Page 233 Configuring Security Features Click Encrypt to encrypt the configuration file(s). Click OK. The target directory will be automatically opened. You can find the encrypted CFG file(s), encrypted key file(s) and an Aeskey.txt file storing plaintext AES key(s). Procedure Encryption method can be configured using the configuration files. Configure the encryption method.
Page 234 Administrator’s Guide for SIP-T2xP IP Phones 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.
Page 235: Upgrading Firmware
SIP-T20P 9.x.x.x.rom Note You can download the latest firmware online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Upgrade via Web User Interface To manually upgrade firmware via web user interface, you need to store the firmware to your local system in advance. To upgrade firmware manually via web user interface: Click on Settings->Upgrade.
Page 236 Administrator’s Guide for SIP-T2xP IP Phones A dialog box pops up to prompt “Firmware of the SIP Phone will be updated. It will take 5 minutes to complete. Please don't power off!”. Click OK to confirm the upgrading. Note Do not unplug the network and power cables when the IP phone is upgrading firmware.
Page 237 Upgrading Firmware Upgrading Firmware on page 368. Configure the way for the IP phone to check for configuration files. Local Web User Interface Navigate to: http://<phoneIPAddress>/servl et?p=settings-autop&q=load To configure the way for the IP phone to check for new configuration files via web user interface: Click on Settings->Auto Provision.
Page 238 Administrator’s Guide for SIP-T2xP IP Phones...
Page 239: Resource Files
IP phones. The resources files can be local contact directory, remote phone book and so on. Ask Yealink field application engineer for resource file templates. If the resource file is to be used for all IP phones of the same model, the resource file access URL is best specified in the <y0000000000xx>.cfg file.
Page 240: Dial-Now Template
Administrator’s Guide for SIP-T2xP IP Phones Procedure Use the following procedures to customize a replace rule template. To customize a replace rule template: Open the template file using an ASCII editor. Add the following string to the template, each starting on a separate line: <Data Prefix=""...
Page 241: Softkey Layout Template
Resource Files Procedure Use the following procedures to customize a dial-now template. To customize a dial-now template: Open the template file using an ASCII editor. Add the following string to the template, each starting on a separate line: <Data DialNowRule="" LineID=""/> Where: DialNowRule=""...
Page 242 Administrator’s Guide for SIP-T2xP IP Phones end of the default soft key list, the default soft keys are displayed on the LCD screen by default. Procedure Use the following procedures to customize a softkey layout template. To customize a softkey layout template: Open the template file using an ASCII editor.
Page 243: Local Contact File
Resource Files Local Contact File 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 244: Remote Xml Phone Book
Administrator’s Guide for SIP-T2xP IP Phones mobile_number="" specifies the mobile number of the contact. other_number="" specifies the other number of the contact. line="" specifies the line you want to add this contact to. ring="" specifies the ring tone for this contact.
Page 245: Specifying The Access Url Of Resource Files
</DirectoryEntry> </YealinkIPPhoneDirectory> Note Yealink supplies a phonebook generation tool to generate a remote XML phone book. For more information, refer to Yealink Phonebook Generation Tool User Guide. Specifying the Access URL of Resource Files Access URL of the resource file can be configured in the configuration files: Configure the access URL of the replace rule template.
Page 246: Local Contact File On
Administrator’s Guide for SIP-T2xP IP Phones the dial-now rule template. For more information, refer to Access URL of Dial-now Template on page 371. Configure the access URL of the softkey layout template. Configuration File <y0000000000xx>.cfg For more information, refer to...
Page 247: Troubleshooting
Troubleshooting This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using SIP-T2xP IP phones. Troubleshooting Methods 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 248 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. A dialog box pops up to prompt “Do you want to restart your machine?”. The configuration will take effect after reboot.
Page 249 Troubleshooting 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 reboot. Click OK to reboot the IP phone. The system log will be exported successfully to the desired syslog server after reboot.
Page 250: Capturing Packets
Administrator’s Guide for SIP-T2xP IP Phones The following figure shows a portion of a log file: Capturing Packets You can capture packet in two ways: capturing the packet via web user interface or using the Ethernet software. You can analyze the packet captured for troubleshooting purpose.
Page 251: Enabling Watch Dog Feature
Troubleshooting Click Export to open the file download window, and then save the file to your local system. To capture packet 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 252: Getting Information From Status Indicators
Administrator’s Guide for SIP-T2xP IP Phones Select the desired value from the pull-down list of Watch Dog. Click Confirm to accept the change. Getting Information from Status Indicators Status indicators may consist of the power LED, MESSAGE key LED, line key indicator, headset key indicator and the on-screen icon or error messages.
Page 253: Troubleshooting Solutions
Troubleshooting Solutions 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. Why is the LCD screen blank? Do one of the following: Ensure that the IP phone is properly plugged into a functional AC outlet.
Page 254: Why Does The Ip Phone Display "No Service
Administrator’s Guide for SIP-T2xP IP Phones Why does the IP phone display “No Service”? The LCD screen prompts “No Service” message when there is no available SIP account on the IP phone. Do one of the following: Ensure that an account is actively registered on the IP phone at the path ...
Page 255: What Is The Difference Between A Remote Phone Book And A Local Phone Book
Troubleshooting jitter, due to message recombination of transmission or receiving equipment (e.g., timeout handling, retransmission mechanism, buffer under run). Noisy equipment, such as a computer or a fan, may cause voice interference. Turn  off any noisy equipment. Line issues can also cause this problem; disconnect the old line and redial the call ...
Page 256: Why Does The Ip Phone Use Dob Format Logo File Instead Of Popular Bmp, Jpg And So On
(the size of the uncompressed file compared to that of the compressed file) and can be stored in less space. Tools for converting BMP format to DOB format are available. For more information, refer to Yealink SIP-T2 Series/T3 Series/VP530 IP Phones Auto Provisioning Guide.
Page 257: Why Doesn't The Ip Phone Update The Configuration
Troubleshooting Why doesn’t the IP phone update the configuration? Do one of the following: Ensure that the configuration is set correctly.  Reboot the IP phone. Some configurations require a reboot to take effect.  Ensure that the configuration is applicable to the IP phone model. ...
Page 258: How To Restore The Administrator Password
Administrator’s Guide for SIP-T2xP IP Phones 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?”. Click OK to confirm the resetting. The phone will be reset to factory sucessfully after startup.
Page 259 Troubleshooting Phone Logo Memory Line Key Browser Model Display and an icon messages line) via web user interface)
Page 260 Administrator’s Guide for SIP-T2xP IP Phones...
Page 261: Appendix
Appendix Appendix Appendix A: Glossary 802.1x--an IEEE Standard for port-based Network Access Control (PNAC). It is part of the IEEE 802.1 group of networking protocols. It provides an authentication mechanism to devices wishing to attach to a LAN or WLAN. ACD (Automatic Call Distribution)--used to distribute calls from large volumes of incoming calls to the registered IP phone users.
Page 262 Administrator’s Guide for SIP-T2xP IP Phones HTTPS (Hypertext Transfer Protocol over Secure Socket Layer)--a widely-used communications protocol for secure communication over a network. IEEE (Institute of Electrical and Electronics Engineers)--a non-profit professional association headquartered in New York City that is dedicated to advancing technological innovation and excellence.
Page 263: Appendix B: Time Zones
Appendix Appendix B: Time Zones Time Zone Time Zone Name −11:00 Samoa −10:00 United States-Hawaii-Aleutian −10:00 United States-Alaska-Aleutian −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...
Page 264 Administrator’s Guide for SIP-T2xP IP Phones Time Zone Time Zone Name United Kingdom(London) Morocco +01:00 Albania(Tirane) +01:00 Austria(Vienna) +01:00 Belgium(Brussels) +01:00 Caicos +01:00 Chad +01:00 Croatia(Zagreb) +01:00 Czech Republic(Prague) +01:00 Denmark(Kopenhagen) +01:00 France(Paris) +01:00 Germany(Berlin) +01:00 Hungary(Budapest) +01:00 Italy(Rome) +01:00...
Page 265 Appendix Time Zone Time Zone Name +05:00 Kazakhstan(Aqtobe) +05:00 Kyrgyzstan(Bishkek) +05:00 Pakistan(Islamabad) +05:00 Russia(Chelyabinsk) +05:30 India(Calcutta) +06:00 Kazakhstan(Astana, Almaty) +06:00 Russia(Novosibirsk, Omsk) +07:00 Russia(Krasnoyarsk) +07:00 Thailand(Bangkok) +08:00 China(Beijing) +08:00 Singapore(Singapore) +08:00 Australia(Perth) +09:00 Korea(Seoul) +09:00 Japan(Tokyo) +09:30 Australia(Adelaide) +09:30 Australia(Darwin) +10:00 Australia(Sydney, Melbourne, Canberra) +10:00...
Page 266: Appendix C: Configuration Parameters
Administrator’s Guide for SIP-T2xP IP Phones Appendix C: Configuration Parameters This appendix describes configuration parameters in the configuration files for each feature. The configuration files are <y0000000000xx>.cfg and <MAC>.cfg. Setting Parameters in Configuration Files You can set parameters in the configuration files to configure IP phones. The <y0000000000xx>.cfg and <MAC>.cfg files are stored on the provisioning server.
Page 267 Appendix Static Network Settings Configuration File Parameter- network.internet_port.type <MAC>.cfg Configures the Internet port type. Note: If you change this parameter, the IP Description phone will reboot to make the change take effect. Format Integer Default Value Valid values are: 0-DHCP Range 1-PPPoE 2-Static IP Address...
Page 268 Administrator’s Guide for SIP-T2xP IP Phones port type is configured as Static IP Address and the IP address mode is configured as IPv4 or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 269 Appendix Range Not Applicable network.internet_port.gateway = Example 192.168.1.254 Parameter- Configuration File network.primary_dns <MAC>.cfg Configures the primary DNS server when the Internet port type is configured as Static IP Address and the IP address mode is configured as IPv4 or IPv4&IPv6. Description Note: If you change this parameter, the IP phone will reboot to make the change take...
Page 270 Administrator’s Guide for SIP-T2xP IP Phones PPPoE Configuration File Parameter- network.internet_port.type <MAC>.cfg Configures the Internet port type. Note: If you change this parameter, the IP Description phone will reboot to make the change take effect. Format Integer Default Value Valid values are:...
Page 271 Appendix Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format String Default Value Blank Range Not Applicable Example network.pppoe.password = yealink123 Internet and PC Ports Transmission Methods Internet Port Transmission Method Parameter- Configuration File network.internet_port.speed_d...
Page 272 Administrator’s Guide for SIP-T2xP IP Phones Default Value Valid values are: 0-Auto negotiate 1-Full duplex, 10Mbps Range 2-Full duplex, 100Mbps 3-Half duplex, 10Mbps 4-Half duplex, 100Mbps Example network.pc_port.speed_duplex = 0 PC Port Mode Parameter- Configuration File network.PC_port.enable <y0000000000xx>.cfg Enables or disables the PC port.
Page 273 Appendix Parameter- Configuration File network.pc_port.ip <y0000000000xx>.cfg Configures the IP address for the PC port when the PC port is configured as Router. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value...
Page 274 Administrator’s Guide for SIP-T2xP IP Phones Example network.pc_port.dhcp_server = 1 Parameter- Configuration File network.dhcp.start_ip <y0000000000xx>.cfg Configures the start IP address that the IP phone assigns for the PC attached to the PC port when the PC port is configured as Router.
Page 275 Appendix Replaced, Line ID Enabled/Disabled: Enables or disables the replace rule. Prefix: Specifies the string you want to replace. Replaced: Specifies the alternate string instead of what the user enters. Line ID: Specifies the desired line to apply this replace rule. The digit 0 stands for all lines. X ranges from 1 to 100.
Page 276 Administrator’s Guide for SIP-T2xP IP Phones comma. Format String, Integer Default Value Blank Dial-now Rules: Not Applicable Valid values of Line ID are: Range 0 to 6 (for T28P) 0 to 3 (for T26P/T20P) 0 to 2 (for T20P) Example dialnow.item.1 = 2216,1,2,3...
Page 277 Appendix Parameter- Configuration File dialplan.area_code.min_len <y0000000000xx>.cfg Configures the minimum length of the entered Description numbers. Integer Format Default Value Range 1 to 15 Example dialplan.area_code.min_len = 1 Parameter- Configuration File dialplan.area_code.max_len <y0000000000xx>.cfg Configures the maximum length of the entered numbers. Description Note: The value must be larger than the minimum length.
Page 278 Administrator’s Guide for SIP-T2xP IP Phones Block Out Configuration File Parameter- dialplan.block_out.number.x <y0000000000xx>.cfg Configures the block out numbers. Description X ranges from 1 to 10. Format String Default Value Blank Range Not Applicable Example dialplan.block_out.number.1 = 1234 Parameter- Configuration File dialplan.block_out.line_id.x...
Page 279 Appendix Default Value Range 1 to 10 Example phone_setting.contrast = 6 Backlight Configuration File Parameter- phone_setting.active_backlight <y0000000000xx>.cfg _level Configures the backlight idle intensity used to adjust the backlight intensity of the LCD screen Description Level 3 is the brightest. Note: It is only applicable to the SIP-T28P IP phone.
Page 280 Administrator’s Guide for SIP-T2xP IP Phones Example phone_setting.backlight_time = 30 User Password Parameter- Configuration File security.user_password <y0000000000xx>.cfg Configures a new user password for the IP phone. The IP phone uses “user” as the default user Description password. Note: IP phones support ASCII characters 32-126(0x20-0x7E) only in passwords.
Page 281 Appendix phone_setting.lock <y0000000000xx>.cfg Configures the type of phone lock. Menu Key: The Menu soft key and MESSAGE key are locked (For T20P, the MENU key is locked). Function Keys: MESSAGE, RD, CONF, HOLD, MUTE, TRAN, OK, X, navigation keys, soft keys, line keys and memory keys are locked (For T22P, CONF, HOLD, MUTE and memory keys do not exist;...
Page 282 Administrator’s Guide for SIP-T2xP IP Phones IP phone is locked, you can use the default password “123” to unlock it. Format Not Applicable Default Value Range 0 to 15 characters Example phone_setting.phone_lock.unlock_pin = 123 Parameter- Configuration File phone_setting.phone_lock.loc <y0000000000xx>.cfg k_time_out Configures the IP phone to automatically lock the keypad after a delay time (in seconds).
Page 283 Appendix Parameter- Configuration File local_time.ntp_server2 <y0000000000xx>.cfg Configures the IP address or the domain name of the secondary NTP server. If the primary NTP Description server is not configured or cannot be accessed, the IP phone will request the time and date from the secondary NTP server. Format IP Address or Domain Name Default Value...
Page 284 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File local_time.time_zone_name <MAC>.cfg Configures the desired time zone name. Description For more available time zone name list, refer Appendix B: Time Zones on page 247. Format String Default Value China(Beijing) Range Not Applicable Example local_time.time_zone_name = China(Beijing)
Page 285 Appendix Parameter- Configuration File local_time.start_time <y0000000000xx>.cfg Configures the time to start DST. If “local_time.dst_time_type” is set to 0 (By Date), use the mapping: MM: 1=Jan, 2=Feb,…, 12=Dec DD:1=the first day in a month,…, 31= the last day in a month HH:0=1am, 1=2am,…, 23=12pm If “local_time.dst_time_type”...
Page 286 Administrator’s Guide for SIP-T2xP IP Phones If “local_time.dst_time_type” is set to 1 (By Week), use the mapping: Month: 1=Jan, 2=Feb,…, 12=Dec Week of Month: 1=the first week in a month,…, 5=the last week in a month Day of Week: 1=Mon, 2=Tues,…, 7=Sun Hour of Day: 0=1am, 1=2am,…, 23=12pm...
Page 287 Appendix hour format. Format Integer Default Value 0-12 Hour Range 1-24 Hour Example local_time.time_format = 1 Date Format Configuration File Parameter- local_time.date_format <y0000000000xx>.cfg Configures the date format. IP phones support various date formats. You Description can change the desired format according to your requirement.
Page 288 Administrator’s Guide for SIP-T2xP IP Phones Note: The language packs you load are dependent on available language packs from the provisioning server. You can download the language pack to the phone user interface only. Format Default Value Blank Range Not Applicable...
Page 289 Appendix user interface depends on the language preferences of your browser. If the language of your browser is not supported by the IP phone, the web user interface will use English by default. Format String Default Value Not Applicable Valid values are: English Deutsch French...
Page 290 Administrator’s Guide for SIP-T2xP IP Phones value is 1. Valid values are: 0-Disabled 1-System logo 2-Custom logo Range Note: For the SIP-T28 IP phone, valid values are 1(System logo) and 2(Custom logo). For the SIP-T20P IP phones, valid values are 0(Disabled) and 1(Enabled).
Page 291 Appendix Key as Send Configuration File Parameter- features.pound_key.mode <y0000000000xx>.cfg Configures the "#" or "*" key as the send key. If set to 0 (Disabled), neither “#” nor “*” can be used as a send key. Description If set to 1(# key), the pound key is used as the send key.
Page 292 Administrator’s Guide for SIP-T2xP IP Phones Hotline Configuration File Parameter- features.hotline_number <y0000000000xx>.cfg Configures the hotline number. It specifies a number that the IP phone automatically dials out when lifting the Description handset, pressing the speakerphone key or the line key. Leaving it blank disables hotline feature.
Page 293 Appendix Call Log Configuration File Parameter- features.history_save_display <y0000000000xx>.cfg Enables or disables the IP phone to display the Save Call Log option on the web user interface. Description If set to 0 (Disabled), the Save Call Log option is hidden on the web user interface. If set to 1 (Enabled), you can enable or disable call log feature via web user interface.
Page 294 Administrator’s Guide for SIP-T2xP IP Phones displaying on the LCD screen, the IP phone does not log the missed call in the Missed Calls list. If set to 1 (Enabled), a prompt message "<number> New Missed Call(s)" along with an indicator icon is displayed on the IP phone idle screen when the IP phone misses calls.
Page 295 Appendix Default Value Range 1 to 14 Example phone_setting.inter_digit_time Call Waiting Configuration File Parameter- call_waiting.enable <y0000000000xx>.cfg Enables or disables call waiting feature. If set to 0 (Disabled), a new incoming call is automatically rejected by the IP phone with a Description busy message while during a call.
Page 296 Administrator’s Guide for SIP-T2xP IP Phones Auto Redial Configuration File Parameter- auto_redial.enable <y0000000000xx>.cfg Enables or disables the IP phone to automatically redial the called number when it is busy. Description If set to 1 (Enabled), the IP phone dials the previous dialed out number automatically when the dialed number is busy.
Page 297 Appendix Range 1 to 300 Example auto_redial.times = 10 Auto Answer Parameter- Configuration File account.x.auto_answer <MAC>.cfg Enables or disables auto answer feature for account x. If set to 1 (Enabled), the IP phone can automatically answer an incoming call. Description X ranges from 1 to 6.
Page 298 Administrator’s Guide for SIP-T2xP IP Phones 1-Enabled Example features.call_completion_enable = 1 Anonymous Call Configuration File Parameter- account.x.anonymous_call <MAC>.cfg Enables or disables anonymous call feature for account x. If set to 1 (Enabled), the IP phone blocks its identity from showing up to the callee when Description placing a call.
Page 299: Anonymous Call On
Appendix account.x.anonymous_call_on <MAC>.cfg code Configures the anonymous call on code to activate the server-side anonymous call feature for account x (optional). Description X ranges from 1 to 6. Note: It works only if the parameter “account.x.send_anonymous_code” is set to 1 (Enabled).
Page 300: Anonymous Call Rejection On
Administrator’s Guide for SIP-T2xP IP Phones automatically rejects incoming calls from users enabled anonymous call feature. The anonymous user’s phone LCD screen presents “Anonymity Disallowed”. X ranges from 1 to 6. Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.reject_anonymous_call = 1...
Page 301 Appendix Do Not Disturb Return Message When DND Parameter- Configuration File features.dnd_refuse_code <y0000000000xx>.cfg Configures return codes and reason of the SIP response message when rejecting an incoming call for DND. A specific reason is displayed on the caller’s phone LCD screen. Description If set to 486 (Busy here), the caller’s phone LCD screen displays the reason “Busy here”...
Page 302 Administrator’s Guide for SIP-T2xP IP Phones DND in Phone Mode Configuration File Parameter- features.dnd.enable <y0000000000xx>.cfg Enables or disables DND feature. Description If set to 1 (Enabled), the IP phone rejects incoming calls on all accounts. Format Boolean Default Value 0-Disabled...
Page 303 Appendix If set to 1 (Enabled), the IP phone rejects incoming calls on account x. X ranges from 1 to 6. Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.dnd.enable = 1 Configuration File Parameter- account.x.dnd.on_code <MAC>.cfg Configures the DND on code to activate the server-side DND feature for account x Description (optional).
Page 304 Administrator’s Guide for SIP-T2xP IP Phones Busy Tone Delay Configuration File Parameter- features.busy_tone_delay <y0000000000xx>.cfg Configures a period of time (in seconds) for which the busy tone is audible on the IP phone. When one party releases the call, a busy tone...
Page 305 Appendix 486-Busy here Example features.normal_refuse_code = 486 180 Ring Workaround Parameter- Configuration File phone_setting.is_deal180 <y0000000000xx>.cfg Enables or disables the IP phone to deal with the 180 SIP message received after the 183 SIP message. Description If set to 1 (Enabled), the IP phone resumes and plays the local ringback tone upon a subsequent 180 message received.
Page 306 Administrator’s Guide for SIP-T2xP IP Phones SIP Session Timer Configuration File Parameter- account.x.advanced.timer_t1 <MAC>.cfg Configures the SIP session timer T1 (in seconds) for account x. T1 is an estimate of the Round Trip Time (RTT) Description of transactions between a SIP client and SIP server.
Page 307 Appendix to clear messages between the SIP Client and SIP Server. X ranges from 1 to 6. Format Float Default Value Range 2.5 to 60 Example account.1.advanced.timer_t4 = 5 Session Timer Parameter- Configuration File account.x.session_timer.enable <MAC>.cfg Enables or disables the session timer for account x.
Page 308 Administrator’s Guide for SIP-T2xP IP Phones Example account.1.session_timer.expires = 1800 Parameter- Configuration File account.x.session_timer.refresher <MAC>.cfg Configures the session timer refresher for account x. If set to 0 (UAC), refreshing the session is performed by the IP phone. Description If set to 1 (UAS), refreshing the session is performed by a SIP server.
Page 309 Appendix tone every 30 seconds when there is a hold call on the IP phone. Note: It works only if the parameter “features.play_hold_tone.enable” is set to 1 (Enabled). Format Integer Default Value Range Not Applicable Example features.play_hold_tone.delay = 30 Parameter- Configuration File sip.rfc2543_hold <y0000000000xx>.cfg...
Page 310 Administrator’s Guide for SIP-T2xP IP Phones forward feature for each account. Format Integer Default Value 0-Phone Range 1-Custom Example features.fwd_mode = 0 Call Forward in Phone Mode Always Forward Parameter- Configuration File forward.always.enable < y0000000000xx >.cfg Enables or disables always forward feature.
Page 311 Appendix Format String Default Value Blank Range Not Applicable Example forward.always.on_code = *72 Parameter- Configuration File forward.always.off_code < y0000000000xx >.cfg Configures the always forward off code to Description deactivate the server-side always forward feature. Format String Default Value Blank Range Not Applicable Example forward.always.off_code = *73...
Page 312 Administrator’s Guide for SIP-T2xP IP Phones Range Not Applicable Example forward.busy.target = 3602 Parameter- Configuration File forward.busy.on_code < y0000000000xx >.cfg Configures the busy forward on code to Description activate the server-side busy forward feature. Format String Default Value Blank Range...
Page 313 Appendix Example forward.no_answer.enable = 1 Parameter- Configuration File forward.no_answer.target < y0000000000xx >.cfg Configures the destination number of the no Description answer forward. Format String Default Value Blank Range Not Applicable Example forward.no_answer.target = 3603 Parameter- Configuration File forward.no_answer.timeout < y0000000000xx >.cfg Configures a period of ring time to wait before forwarding the incoming call.
Page 314 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File forward.no_answer.off_code < y0000000000xx >.cfg Configures the no answer forward off code Description to deactivate the server-side no answer forward feature. Format String Default Value Blank Range Not Applicable Example forward.no_answer.off_code = *77...
Page 315 Appendix Parameter- Configuration File account.x.always_fwd.on_code <MAC>.cfg Configures the always forward on code activate the server-side always forward Description feature for account x. X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example account.1.always_fwd.on_code = *72 Parameter- Configuration File account.x.always_fwd.off_code...
Page 316 Administrator’s Guide for SIP-T2xP IP Phones Example account.1.busy_fwd.enable = 1 Parameter- Configuration File account.x.busy_fwd.target <MAC>.cfg Configures the destination number of the busy forward for account x. Description X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example account.1.busy_fwd.target = 3602...
Page 317 Appendix No Answer Forward Configuration File Parameter- account.x.timeout_fwd.enable <MAC>.cfg Enables or disables no answer forward feature for account x. If set to 1 (Enabled), incoming calls to the Description account x are forward to the destination number after a period of ring time. X ranges from 1 to 6.
Page 318 Administrator’s Guide for SIP-T2xP IP Phones Range 0 to 20 Example account.1.timeout_fwd.timeout = 2 Parameter- Configuration File account.x.timeout_fwd.on_code <MAC>.cfg Configures the no answer forward on code to activate the server-side no answer Description forward feature for account x. X ranges from 1 to 6.
Page 319 Appendix Example forward.international.enable = 1 Call Transfer Parameter- Configuration File transfer.blind_tran_on_hook_ena <y0000000000xx>.cfg Enables or disables the IP phone to complete Description the blind transfer through on-hook. Format Boolean Default Value 0-Disabled Range 1-Enabled Example transfer.blind_tran_on_hook_enable = 1 Configuration File Parameter- transfer.on_hook_trans_enable <y0000000000xx>.cfg Enables or disables the IP phone to complete...
Page 320 Administrator’s Guide for SIP-T2xP IP Phones Example transfer.semi_attend_tran_enable = 1 Network Conference Parameter- Configuration File account.x.conf_type <MAC>.cfg Configures the conference type for account If set to 0 (Local Conference), conferences are set up on the IP phone locally. Description If set to 2 (Network Conference), conferences are set up by the server.
Page 321 Appendix Transfer on Conference Hang Up Configuration File Parameter- transfer.tran_others_after_conf_e <y0000000000xx>.cfg nable Enables or disables Transfer on Conference Hang Up feature. If enabled, the other two parties remain Description connected when the conference initiator drops the conference call. Note: It is only applicable to the local conference.
Page 322 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File features.pickup.direct_pickup_c <y0000000000xx>.cfg Configures the directed call pickup code on a phone basis. Note: The directed call pickup code Description configured on a per-line basis takes precedence over that configured on a phone basis.
Page 323 Appendix the GPickup soft key when the IP phone is off-hook. Format Boolean Default Value 0-Disabled Range 1-Enabled Example features.pickup.group_pickup_enable = 1 Parameter- Configuration File features.pickup.group_pickup_c <y0000000000xx>.cfg Configures the group call pickup code on a phone basis. Note: The group call pickup code Description configured on a per-line basis takes precedence over that configured on a...
Page 324 Administrator’s Guide for SIP-T2xP IP Phones Dialog-Info Call Pickup Configuration File Parameter- account.x.dialoginfo_callpickup <MAC>.cfg Configures Dialog-Info Call Pickup feature for account x. Description If set to 1 (Enabled), call pickup is implemented through SIP signals. X ranges from 1 to 6.
Page 325 Appendix phone will reboot to make the change take effect. Format Integer Default Value Range 1 to 65535 Example network.port.http = 80 Parameter- Configuration File wui.https_enable <y0000000000xx>.cfg Enables or disables the IP phone to access its web user interface using HTTPS protocol. Description Note: If you change this parameter, the IP phone will reboot to make the change take...
Page 326 Administrator’s Guide for SIP-T2xP IP Phones Calling Line Identification Presentation Configuration File Parameter- account.x.cid_source <MAC>.cfg Configures the presentation of the caller identity for account x. 0-FROM (Derives the name and number of the caller from the “From” header). 1-PAI (Derives the name and number of the caller from the “PAI”...
Page 327 Appendix 2-RFC 4916 (Derives the name and number of the callee from “From” header in the Update message). When the RFC 4916 is enabled on the IP phone, the caller sends the SIP request message which contains the from-change tag in the Supported header. The caller then receives an UPDATE message from the callee, and displays the identity in the From header.
Page 328 Administrator’s Guide for SIP-T2xP IP Phones 2-SIP INFO 3-AUTO or SIP INFO Example account.1.dtmf.type = 1 Parameter- Configuration File account.x.dtmf.dtmf_payload <MAC>.cfg Configures the RFC 2833 payload type. Description X ranges from 1 to 6. Format Integer Default Value Range 96 to 127 Example account.1.dtmf.dtmf_payload = 101...
Page 329 Appendix Suppress DTMF Display Configuration File Parameter- features.dtmf.hide <y0000000000xx>.cfg Enables or disables the IP phone to suppress the display of DTMF digits. Description If set to 1 (Enabled), the DTMF digits are displayed as asterisks. Format Boolean Default Value 0-Disabled Range 1-Enabled Example...
Page 330 Administrator’s Guide for SIP-T2xP IP Phones the specified DTMF digits to the server for completing call transfer when pressing the transfer key during a call. Format Boolean Default Value 0-Disabled Range 1-Enabled Example features.dtmf.replace_tran = 1 Configuration File Parameter- features.dtmf.transfer <y0000000000xx>.cfg...
Page 331 Appendix 0-Disabled Range 1-Enabled Example features.intercom.allow = 1 Parameter- Configuration File features.intercom.mute <y0000000000xx>.cfg Enables or disables the IP phone to mute the microphone when answering an intercom call. Description If set to 0 (Disabled), the microphone is un-muted for incoming calls. If set to 1 (Enabled), the microphone is muted for intercom calls.
Page 332 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File features.intercom.barge <y0000000000xx>.cfg Enables or disables the IP phone to automatically answer an incoming intercom call while there is already an active call on the IP phone. If set to 0 (Disabled), the IP phone handles an...
Page 333 Appendix X ranges from 1 to 6. Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.alert_info_url_enable = 1 Configuration File Parameter- distinctive_ring_tones.alert_info.x <y0000000000xx>.cfg .text Configures the texts to map the keywords contained in the SIP header. Description X ranges from 1 to 10. Format String Default Value...
Page 334 Administrator’s Guide for SIP-T2xP IP Phones Tones Configuration File Parameter- voice.tone.country <y0000000000xx>.cfg Description Configures the country tone for the IP phone. Format String Default Value Custom Valid values are: Custom  Australia  Austria  Brazil  Belgium  China ...
Page 335 Appendix voice.tone.ring <y0000000000xx>.cfg voice.tone.busy voice.tone.congestion voice.tone.callwaiting voice.tone.dialrecall voice.tone.info voice.tone.stutter voice.tone.message voice.tone.autoanswer Configures the tone for each condition. tonelist = element[,element] [,element]… Where element = [!]freq1[+freq2][+freq3][+freq4] /duration Freq: the frequency of the tone (ranges from 200 to 7000 Hz). If set to 0 (0Hz), it means the tone is not played.
Page 336 Administrator’s Guide for SIP-T2xP IP Phones remote_phonebook.data.x.url <y0000000000xx>.cfg Configures the access URL of the remote XML phone book. Description X ranges from 1 to 5. Format Default Value Blank Range Not Applicable remote_phonebook.data.1.url = Example http://192.168.1.20/phonebook.xml Parameter- Configuration File remote_phonebook.data.x.nam <y0000000000xx>.cfg...
Page 337 Appendix Configures how often to refresh the local cache of the remote phone book. Description If set to 3600 (3600s), the IP phone refreshes the local cache of the remote phone book every 3600 seconds. Format Integer Default Value 21600 Range 120 to 2592000 features.remote_phonebook.flash_time =...
Page 338 Administrator’s Guide for SIP-T2xP IP Phones Format String Default Value Blank Range Not Applicable ldap.number_filter = (|(telephoneNumber=%)(Mobile=%)(ipPh one=%)) When the number prefix of the Example telephoneNumber, Mobile or ipPhone of the contact record matches the search criteria, the record will be displayed on the LCD screen.
Page 339 Format String Default Value Blank Range Not Applicable Example ldap.base = dc=yealink,dc=cn Parameter- Configuration File ldap.user <y0000000000xx>.cfg Configures the user name uses to login the LDAP server. This parameter can be left blank in case the Description server allows anonymous to login.
Page 340 Administrator’s Guide for SIP-T2xP IP Phones Default Value Blank Range Not Applicable Example ldap.password = secret Parameter- Configuration File ldap.max_hits <y0000000000xx>.cfg Configures the maximum number of search results to be returned by the LDAP server. If the value of the “Max.Hits” is blank, the LDAP server will return all searched results.
Page 341 Appendix configure multiple number attributes separated by space. Format String Default Value Blank Range Not Applicable Example ldap.numb_attr = telephoneNumber Parameter- Configuration File ldap.display_name <y0000000000xx>.cfg Configures the display name of the contact record displayed on the LCD screen. Description Note: It must start with “%” symbol. Format String Default Value...
Page 342 Administrator’s Guide for SIP-T2xP IP Phones call. Format Boolean Default Value 0-Disabled Range 1-Enabled Example ldap.call_in_lookup = 1 Parameter- Configuration File ldap.ldap_sort <y0000000000xx>.cfg Enables or disables the IP phone to sort the Description search results in alphabetical order or numerical order.
Page 343 Configuration File account.x.music_server_uri <MAC>.cfg Configures the Music on Hold server address. Examples for valid values: <10.1.3.165>, 10.1.3.165, sip:moh@sip.com, <sip:moh@sip.com>, <yealink.com> or Description yealink.com. X ranges from 1 to 6. Note: The DNS query in this parameter only supports A query.
Page 344 Administrator’s Guide for SIP-T2xP IP Phones Range Not Applicable Example account.1.music_server_uri =<10.1.3.165> Parameter- Configuration File account.x.acd.enable <MAC>.cfg Enables or disables ACD feature for account Description X ranges from 1 to 6. Format Boolean Default Value 0-Disabled Value 1-Enabled Example account.1.acd.enable = 1...
Page 345 Appendix Format Boolean Default Value 0-Disabled Value 1-Enabled Example acd.auto_available = 1 Parameter- Configuration File acd.auto_available_timer <y0000000000xx>.cfg Configures the length of time (in seconds) before the IP phone state is automatically changed to available. Description Note: It works only if the parameter “acd.auto_available”...
Page 346 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File account.x.subscribe_mwi_expires <MAC>.cfg Configures MWI subscribe expiry time (in seconds) for account x. The IP phone is able to successfully refresh the SUBCRIBE for message-summary events before expiration of the SUBSCRIBE dialog.
Page 347 Appendix Format Boolean Default Value 0-Disabled Value 1-Enabled Example account.1.subscribe_mwi_to_vm = 0 Sending RTP Stream Parameter- Configuration File multicast.codec <y0000000000xx>.cfg Configures a multicast codec for the IP Description phone to use to send an RTP stream. Format string Default Value G722 Valid values are: PCMU...
Page 348 Administrator’s Guide for SIP-T2xP IP Phones Format Boolean Default Value 0-Disabled Range 1-Enabled Example multicast.receive_priority.enable =1 Configuration File Parameter- multicast.receive_priority.priority < y0000000000xx >.cfg Configures the priority of multicast paging calls. 1 is the highest priority, 10 is the lowest Description priority.
Page 349 Appendix number that the IP phone listens to. X ranges from 1 to 10. Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255. Format String Default Value Blank Range Not Applicable multicast.listen_address.1.ip_address = Example 224.5.6.20:10008 Action URL Parameter- Configuration File action_url.setup_completed <y0000000000xx>.cfg...
Page 350 Administrator’s Guide for SIP-T2xP IP Phones action_url.unmute action_url.missed_call action_url.call_terminated action_url.busy_to_idle action_url.idle_to_busy action_url.ip_change action_url.forward_incoming_call action_url.reject_incoming_call action_url.answer_new_incoming_ call action_url.transfer_finished action_url.transfer_failed Configures the URL for the predefined event. The value format is: http(s)://IP address of server/help.xml? variable name=variable value. Valid variable values are: $mac ...
Page 351 Appendix Action URI Configuration File Parameter- features.action_uri_limit_ip <y0000000000xx>.cfg Configures the address(es) from which Action URI will be accepted. For discontinuous IP addresses, each IP address is separated by comma. For continuous IP addresses, the format likes *.*.*.* and the “*” stands for the values 0~255.
Page 352 Administrator’s Guide for SIP-T2xP IP Phones yealink.pbx.com Configuration File Parameter- account.x.sip_server.y.port <MAC>.cfg Configures the port of the SIP server for account x. Description X ranges from 1 to 6. Y ranges from 1 to 2. Format Integer Default Value 5060...
Page 353: Failover Mode
Appendix Example account.1.sip_server.1.retry_counts = 3 Fallback Mode Parameter- Configuration File account.x.fallback.redundancy_ty <MAC>.cfg Configures the registration mode for the IP phone in fallback mode. Description X ranges from 1 to 6. Format Integer Default Value Valid values are: Range 0-Concurrent registration 1-Successive registration account.1.fallback.redundancy_type = Example...
Page 354 Administrator’s Guide for SIP-T2xP IP Phones Configures the way in which the phone fails back to the primary server for call control in the failover mode. Description X ranges from 1 to 6. Y ranges from 1 to 2. Format...
Page 355 Appendix X ranges from 1 to 6. Y ranges from 1 to 2. Format Integer Default Value 3600 Range 0, 60 to 65535 account.1.sip_server.1.failback_timeout = Example 3600 Parameter- Configuration File account.x.sip_server.y.register_on_ <MAC>.cfg enable Enables or disables the IP phone to register to the secondary server before sending requests to the secondary server in the Description...
Page 356 Administrator’s Guide for SIP-T2xP IP Phones Default Value Valid values are: 0-UDP Range 1-TCP 2-TLS 3-DNS-NAPTR Example account.1.transport = 3 Parameter- Configuration File account.x.naptr_build <MAC>.cfg Configures UDP SRV query or TCP/TLS SRV query for the IP phone to be performed...
Page 357 Appendix Parameter- Configuration File network.lldp.packet_interval <y0000000000xx>.cfg Configures the amount of time (in seconds) between the transmissions of LLDP packet. Note: If you change this parameter, the IP Description phone will reboot to make the change take effect. It works only if the parameter “network.lldp.enable”...
Page 358 Administrator’s Guide for SIP-T2xP IP Phones Format Integer Default Value Range 1 to 4094 Example network.vlan.internet_port_vid = 1 Parameter- Configuration File network.vlan.internet_port_priority <y0000000000xx>.cfg Configures the priority value used for passing VLAN packets. 7 is the highest priority, 0 is the lowest Description priority.
Page 359 Appendix Parameter- Configuration File network.vlan.pc_port_vid <y0000000000xx>.cfg Configures the VLAN ID that is associated with the particular VLAN. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value Range 1 to 4094 Example network.vlan.pc_port_vid = 1...
Page 360 Administrator’s Guide for SIP-T2xP IP Phones Example network.vlan.dhcp_enable = 1 Parameter- Configuration File network.vlan.dhcp_option <y0000000000xx>.cfg Configures the DHCP option used to Description request the VLAN ID. Format String Default Value Range 128 to 254 Example network.vlan.dhcp_option = 132 Configuration File Parameter- network.vpn_enable...
Page 361 Appendix Configuration File Parameter- network.qos.rtptos <y0000000000xx>.cfg Configures the DSCP for voice packets. The default DSCP value for RTP packets is 46 (Expedited Forwarding). Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value...
Page 362 Administrator’s Guide for SIP-T2xP IP Phones Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.nat.nat_traversal = 0 Parameter- Configuration File account.x.nat.stun_server <MAC>.cfg Configures the IP address or the domain name of the STUN server for account x. Description X ranges from 1 to 6.
Page 363 Appendix Format Boolean Default Value 0-Disabled Range 1-Enabled Example network.snmp.enable = 1 Configuration File Parameter- network.snmp.port <y0000000000xx>.cfg Configures the port used for SNMP communication. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 364 Administrator’s Guide for SIP-T2xP IP Phones 802.1X Configuration File Parameter- network.802_1x.mode <y0000000000xx>.cfg Configures the types of the 802.1X authentication to use on the IP phone. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 365 Appendix phone will reboot to make the change take effect. It is only applicable to EAP-MD5, PEAP-MSCHAPv2 and EAP-TTLS/EAP-MSCHAPv2 protocols. Format String Default Value Blank Range Not Applicable network.802_1x.md5_password = Example admin123 Parameter- Configuration File network.802_1x.root_cert_url <y0000000000xx>.cfg Configures the access URL of the root certificate used for authentication.
Page 366 Administrator’s Guide for SIP-T2xP IP Phones Default Value Blank Range Not Applicable network.802_1x.client_cert_url = Example http://192.168.1.10/ client.pem TR-069 Configuration File Parameter- managementserver.enable <y0000000000xx>.cfg Enables or disables TR-069 feature on the IP phone. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 367 Appendix with the ACS. This string is set to the empty string if no authentication is required. Note: If you change this parameter, the phone will reboot to make the change take effect. Format String Default Value Blank Range Not Applicable Example managementserver.password = pwd123 Configuration File...
Page 368 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File managementserver.connection_r <y0000000000xx>.cfg equest_password Configures the password for the IP phone to authenticate the incoming connection requests. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 369 Appendix phone will reboot to make the change take effect. Format Integer Default Value Range Not Applicable managementserver.periodic_inform_interv Example al = 60 IPv6 Configuration File Parameter- network.ip_address_mode <MAC>.cfg Configures the IP address mode. Note: If you change this parameter, the IP Description phone will reboot to make the change take effect.
Page 370 Administrator’s Guide for SIP-T2xP IP Phones Example network.ipv6_internet_port.type = 0 Parameter- Configuration File network.ipv6_internet_port.ip <MAC>.cfg Configures the IPv6 address when the IPv6 address assignment method is configured as Static IP Address and the IP address mode is configured as IPv6 or Description IPv4&IPv6.
Page 371 Appendix configured as Static IP Address and the IP address mode is configured as IPv6 or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank Range Not Applicable network.ipv6_internet_port.gateway =...
Page 372: Audio Feature Parameters
Administrator’s Guide for SIP-T2xP IP Phones Format IP Address Default Value Blank Range Not Applicable network.ipv6_secondary_dns = Example 2026:1234:1:1:c3c7:c11c:5447:23a6 Configuration File Parameter- network.ipv6_icmp_v6.enable <MAC>.cfg Enables or disables ICMPv6 feature. If it is set to 1 (enabled), the IP phone obtains network settings of the IPv6 from Description the ICMPv6 protocol.
Page 373 Appendix 0-Disabled Range 1-Enabled Example features.headset_prior = 1 Dual Headset Configuration File Parameter- features.headset_training <y0000000000xx>.cfg Enables or disables dual headset feature. If set to 1 (Enabled), users can use two headsets on one phone. When the IP phone joins in a cal, the users with the Description headset connected to the headset jack have a full-duplex conversation, while the...
Page 374 Administrator’s Guide for SIP-T2xP IP Phones When Y=7, the default value is 0; When Y=8, the default value is 0; When Y=9, the default value is 0; When Y=10, the default value is 0; When Y=11, the default value is 0.
Page 375 Appendix account.1.codec.1.payload_type = Example PCMU Parameter- Configuration File account.x.codec.y.priority <MAC>.cfg Configures the priority for the codec. Description X ranges from 1 to 6. Y ranges from 1 to 11. Format Integer When Y=1, the default value is 1; When Y=2, the default value is 2; When Y=3, the default value is 0;...
Page 376 Administrator’s Guide for SIP-T2xP IP Phones When Y=6, the default value is 9; When Y=7, the default value is 102; When Y=8, the default value is 112; When Y=9, the default value is 102; When Y=10, the default value is 99;...
Page 377 Appendix Voice Activity Detection Configuration File Parameter- voice.vad <y0000000000xx>.cfg Enables or disables VAD feature on the IP Description phone. Format Boolean Default Value 0-Disabled Range 1-Enabled Example voice.vad = 1 Comfort Noise Generation Parameter- Configuration File voice.cng <y0000000000xx>.cfg Enables or disables CNG feature on the IP Description phone.
Page 378 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File voice.jib.min <y0000000000xx>.cfg Configures the minimum delay time for jitter buffer. Description Note: It works only if the parameter “voice.jib.adaptive” is set to 1 (Adaptive). Format Integer Default Value Range Not Applicable Example voice.jib.min = 60...
Page 379: Security Feature Parameters
Appendix Security Feature Parameters Parameter- Configuration File account.x.transport <MAC>.cfg Configures the transport type for account If set to 2 (TLS), the SIP message of this Description account will be encrypted after the successful TLS negotiation. X ranges from 1 to 6. Format Integer Default Value...
Page 380 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File security.ca_cert <y0000000000xx>.cfg Configures the type of certificates the IP phone used to authenticate the connecting server. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Page 381 Appendix Format Boolean Default Value 0-Default certificates Range 1-Custom certificates Example security.dev_cert = 0 Uploading Certificates Parameter- Configuration File trusted_certificates.url <y0000000000xx>.cfg Configures the access URL of the certificate used to authenticate the connecting server. Description Note: The certificate you want to upload must be in *.pem, *.crt, *.cer or *.der format.
Page 382 Administrator’s Guide for SIP-T2xP IP Phones SRTP Configuration File Parameter- account.x.srtp_encryption <MAC>.cfg Configures whether to use voice encryption service. If the set to 1 (Optional), the IP phone will negotiate with the other IP phone what Description type of encryption to utilize for the session.
Page 383 Appendix 0-Disabled Value 1-Enabled Example auto_provision.aes_key_in_file = 0 Parameter- Configuration File auto_provision.aes_key_16.com <y0000000000xx>.cfg Configures the plaintext AES key which is used to decrypt the <y0000000000xx>.cfg file. Description Note: It works only if the parameter “auto_provision.aes_key_in_file” is set to 0 (Disabled). Format String Default Value...
Page 384: Upgrading Firmware
Administrator’s Guide for SIP-T2xP IP Phones Upgrading Firmware Configuration File Parameter- auto_provision.mode <y0000000000xx>.cfg Description Configures the auto provision mode. Format Integer Default Value Valid values are: 0-Disabled 1-Power on (when the IP phone reboots) Range 4-Repeatedly (at a fixed interval)
Page 385 Appendix Note: It works only if the parameter “auto_provision.mode” is set to 5(Weekly) or 7 (Power on + Weekly). Format 00:00 Default Value 00:00 Range 00:00 to 23:59 auto_provision.schedule.time_from = Example 01:30 Parameter- Configuration File auto_provision.schedule.time_to < y0000000000xx >.cfg Configures the end time of day in 24-hour period for the IP phone to check new configuration files.
Page 386: Resource Files
Administrator’s Guide for SIP-T2xP IP Phones 3-Wednesday 4-Thursday 5-Friday 6-Saturday auto_provision.schedule.dayofweek = Example 0123456 Configuration File Parameter- firmware.url <y0000000000xx>.cfg Description Configures the access URL of the firmware. Format String Default Value Blank Range Not Applicable firmware.url = Example http://192.168.1.20/2.71.0.140.rom Resource Files...
Page 387 Appendix Access URL of Dial-now Template Configuration File Parameter- dialplan_dialnow.url <y0000000000xx>.cfg Configures the access URL of the dial-now Description template. Format Default Value Blank Range Not Applicable dialplan_dialnow.url = Example http://192.168.10.25/dialnow.xml Access URL of Softkey Layout Template Configuration File Parameter- custom_softkey_call_failed.url <y0000000000xx>.cfg Configures the access URL of the...
Page 388 Administrator’s Guide for SIP-T2xP IP Phones Format Default Value Not Applicable Range Not Applicable The following example uses HTTP to download the CallIn state file from the “XMLfiles” directory on provisioning Example server 10.2.8.16 using 8080 port. custom_softkey_call_in.url = http://10.2.8.16:8080/XMLfiles/CallIn.xml...
Page 389 Appendix The following example uses HTTP to download the Dialing state file from the “XMLfiles” directory on provisioning Example server 10.2.8.16 using 8080 port. custom_softkey_dialing.url = http://10.2.8.16:8080/XMLfiles/Dialing.xml Configuration File Parameter- custom_softkey_ring_back.url <y0000000000xx>.cfg Configures the access URL of the customized file for the soft key presented Description on the LCD screen when in the RingBack state.
Page 390: Troubleshooting
Administrator’s Guide for SIP-T2xP IP Phones custom_softkey_talking.url = http://10.2.8.16:8080/XMLfiles/Talking.xml Access URL of Local Contact File Parameter- Configuration File local_contact.data.url <y0000000000xx>.cfg Configures the access URL of the local Description contact file. Format Default Value Blank Range Not Applicable local_contact.data.url = Example http://192.168.10.25/contactData1.xml...
Page 391 Appendix server where to export the log files. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank Range Not Applicable Example syslog.server = 192.168.1.50 Configuration File Parameter- syslog.log_level <y0000000000xx>.cfg...
Page 392: Configuring Dss Key
Administrator’s Guide for SIP-T2xP IP Phones Configuring DSS Key This section provides the DSS key parameters you can configure on the IP phone. DSS key consists of memory key and line key. The following table lists the number of DSS keys...
Page 393 Appendix URL (not applicable to SIP-T20P)  Group Listening  XML Group (not applicable to SIP-T20P)  Group Pickup  Multicast Paging  Record  XML Browser (not applicable to SIP-T20P)  URL Record  LDAP (not applicable to SIP-T20P) ...
Page 394 Administrator’s Guide for SIP-T2xP IP Phones 27-XML Browser 34-Hot Desking 35-URL Record 38-LDAP 40-Prefix 41-Zero Touch 42-ACD 45-Local Group 48-Custom Button 50-Keypad Lock 61-Directory Example memorykey.1.type = 8 Parameter- Configuration File memorykey.x.line <y0000000000xx>.cfg Parameter- Line key. x. line Configures the desired line to apply the key feature.
Page 395 Appendix  Hot Desking  Zero Touch  URL (not applicable to the SIP-T20P IP  phone) Keypad Lock  Directory  Format Integer For the memory 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 396 Administrator’s Guide for SIP-T2xP IP Phones the number you want to dial out. memorykey.1.value = 1001 Parameter- Configuration File memorykey.x.pickup_value <y0000000000xx>.cfg Parameter- linekey.x.pickup_value Configures the pickup code for BLF feature. This parameter is only applicable to BLF feature. Description For the memory key, x ranges from 1 to 10.
Page 397 Appendix … Format Integer Default Value Range Not Applicable Specify the second remote phone book. Example memorykey.1.xml_phonebook = 1 Keypad Lock Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Parameter- linekey.x.type Configures a DSS key to be Keypad Lock key on the IP phone. The digit 50 stands for the key type Keypad Description Lock.
Page 398 Administrator’s Guide for SIP-T2xP IP Phones Value Example memorykey.1.type = 5 Directed Call Pickup Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Parameter- linekey.x.type Configures a DSS key to be directed call pickup key on the IP phone. The digit 9 stands for the key type Call Pickup.
Page 399 Appendix 6-Line 6 Example memorykey.1.line = 1 Parameter- Configuration File memorykey.x.value <y0000000000xx>.cfg Parameter- linekey.x.value Configures the directed call pickup feature code followed by the number of monitored extension. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format String Range...
Page 400 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File memorykey.x.line <y0000000000xx>.cfg Parameter- linekey.x.line Configures the desired line to apply the group call pickup key. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Page 401 Appendix Call Return Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Parameter- linekey.x.type Configures a DSS key to be call return key on the IP phone. The digit 7 stands for the key type Call Return. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Page 402 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File memorykey.x.line <y0000000000xx>.cfg Parameter- linekey.x.line Configures the desired line to apply key feature. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Page 403: Intercom Key
Appendix Intercom Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Parameter- linekey.x.type Configures a DSS key to be the intercom key. The digit 14 stands for the key type Intercom. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Value...
Page 404 Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File memorykey.x.value <y0000000000xx>.cfg Parameter- linekey.x.value Configures the intercom number. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format String...
Page 405 Appendix For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Value Example memorykey.3.type = 16 Parameter- Configuration File memorykey.x.line <y0000000000xx>.cfg Parameter- linekey.x.line Configures the desired line to apply the BLF key. Description For the memory key, x ranges from 1 to 10.
Page 406 Administrator’s Guide for SIP-T2xP IP Phones Example memorykey.3.value = 1008 Parameter- Configuration File memorykey.x.pickup_value <y0000000000xx>.cfg Parameter- linekey.x.pickup_value Configures the pickup code for the BLF feature. This parameter only applies to the BLF feature. Description For the memory key, x ranges from 1 to 10.
Page 407 Appendix Multicast Paging Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Parameter- linekey.x.type Configures a DSS key to be a multicast paging key on the IP phone. The digit 24 stands for the key type Multicast Description Paging. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Page 408 Administrator’s Guide for SIP-T2xP IP Phones Record Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Parameter- linekey.x.type Configures a DSS key to be a record key on the IP phone. The digit 25 stands for the key type Record. Description For the memory key, x ranges from 1 to 10.
Page 409: Appendix D: Sip (Session Initiation Protocol)
Example memorykey.2.type = 34 Appendix D: SIP (Session Initiation Protocol) This section describes how Yealink SIP-T2xP 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 410: Rfc And Internet Draft Support
Administrator’s Guide for SIP-T2xP IP Phones RFC and Internet Draft Support The following RFC’s and Internet drafts are supported: RFC 1321—The MD5 Message-Digest Algorithm  RFC 2327—SDP: Session Description Protocol  RFC 2387—The MIME Multipart / Related Content-type  RFC 2976—The SIP INFO Method ...
Page 411: Sip Request
RFC number. SIP Request The following SIP request messages are supported: Method Supported Notes REGISTER Yealink SIP-T2xP IP phones support mid-call changes such as placing a call on INVITE hold as signaled by a new INVITE that contains an existing Call-ID.
Page 412: Sip Header
Administrator’s Guide for SIP-T2xP IP Phones Method Supported Notes UPDATE PUBLISH SIP Header The following SIP request headers are supported: Method Supported Notes Accept Alert-Info Allow Allow-Events Authorization Call-ID Call-Info Contact Content-Length Content-Type CSeq Diversion Event Expires From Max-Forwards Min-SE...
Page 413: Sip Responses
Appendix Method Supported Notes Refer-To Referred-By Remote-Party-ID Replaces Require Route RSeq Session-Expires Subscription-State Supported User-Agent SIP Responses The following SIP responses are supported: 1xx Response—Information Responses 1xx Response Supported Notes 100 Trying 180 Ringing 181 Call Is Being Forwarded 183 Session Progress 2xx Response—Successful Responses 2xx Response Supported...
Page 414 Administrator’s Guide for SIP-T2xP IP Phones 3xx Response—Redirection Responses 3xx Response Supported Notes 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 4xx Response—Request Failure Responses 4xx Response Supported Notes 400 Bad Request 401 Unauthorized 402 Payment Required 403 Forbidden...
Page 415: Sip Session Description Protocol (Sdp) Usage
Appendix 4xx Response Supported Notes 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous 486 Busy Here 487 Request Terminated 488 Not Acceptable Here 491 Request Pending 493 Undecipherable 5xx Response—Server Failure Responses 5xx Response Supported Notes 500 Internal Server Error 501 Not Implemented 502 Bad Gateway...
Page 416: Appendix E: Sip Call Flows
Administrator’s Guide for SIP-T2xP IP Phones o—Owner/creator and session identifier a—Media attribute c—Connection information m—Media name and transport address s—Session name t—Active time Appendix E: SIP Call Flows SIP uses six request methods: INVITE—Indicates a user is being invited to participate in a call session.
Page 417: Successful Call Setup And Disconnect
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 418 Administrator’s Guide for SIP-T2xP IP Phones 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 419: Unsuccessful Call Setup-Called User Is Busy
The following figure illustrates the scenario of an unsuccessful call due to the reason of the called user 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 420 Administrator’s Guide for SIP-T2xP IP Phones 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.
Page 421 Appendix Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
Page 422 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description The proxy server forwards the 486 Busy 486 Busy Here—Proxy Server Here response to notify User A that User to User A B is busy. User A sends a SIP ACK to the proxy server.
Page 423: Unsuccessful Call Setup-Called User Does Not Answer
The following figure illustrates the scenario of an unsuccessful call due to the reason of the called user not answering the 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 424 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 425 Appendix Step Action Description User A wants to disconnect the call. User B sends a SIP 200 OK response to 200 OK—User B to Proxy Server the proxy server. The SIP 200 OK response indicates that User B has received the CANCEL request. The proxy server forwards the SIP 200 OK 200 OK—Proxy Server to User response to notify User A that the...
Page 426: Successful Call Setup And Call Hold
Successful Call Setup and Call Hold 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.
Page 427 Appendix Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
Page 428: Successful Call Setup And Call Waiting
200 OK response. Successful Call Setup and Call Waiting The following figure illustrates a successful call between Yealink SIP IP phones in which parties are in a call, one of the participants receives a call from a third party, then...
Page 429 Appendix 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. User B answers the call. User C calls User B.
Page 430 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 431 Appendix Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A has ACK—User A to Proxy Server received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to User B.
Page 432 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends a mid-call INVITE request to the proxy server with new SDP session INVITE—User A to Proxy Server parameters, which are used to place the call on hold. The proxy server forwards the mid-call INVITE—Proxy Server to User B...
Page 433: Call Transfer Without Consultation
This is called a blind transfer. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
Page 434 Administrator’s Guide for SIP-T2xP IP Phones 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 435 Appendix Step Action Description User A sends an INVITE message to the proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
Page 436 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A has ACK—User A to Proxy Server received the 200 OK response. The call session is now active.
Page 437: Call Transfer With Consultation
This is called attended transfer. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
Page 438 Administrator’s Guide for SIP-T2xP IP Phones Call is established between User B and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7.
Page 439 Appendix Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
Page 440 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A has ACK—User A to Proxy Server received the 200 OK response. The call session is now active.
Page 441 Appendix Step Action Description sends the INVITE request to User C. User C sends a SIP 180 Ringing response 180 Ringing—User C to Proxy to the proxy server. The 180 Ringing Server response indicates that the user is being alerted. The proxy server forwards the 180 180 Ringing—Proxy Server to Ringing response to User A.
Page 442 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description response indicates that User B accepts the transfer. User A terminates the call session by sending a SIP BYE request to the proxy BYE—User A to Proxy Server server. The BYE request indicates that User A wants to release the call.
Page 443: Always Call Forward
User C when User A calls User B. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
Page 444 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 the User B is ...
Page 445 Appendix Step Action Description User A sends a SIP INVITE request to the proxy server. In the INVITE request, a INVITE—User A to Proxy Server unique Call-ID is generated and the Contact-URI field indicates that User A requested the call. The proxy server maps the SIP URI in the INVITE—Proxy Server to User C To field to User C.
Page 446: Busy Call Forward
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 447 Appendix Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
Page 448 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description message. 302 Move Temporarily—Proxy The proxy server forwards the 302 Server to User A Moved Temporarily message to User A. User A sends a SIP ACK to the proxy server. The ACK message notifies the ACK—User A to Proxy Server...
Page 449: No Answer Call Forward
User C when User B does not answer the incoming call after a period of time. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
Page 450 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
Page 451 Appendix Step Action Description message. 302 Move Temporarily—Proxy The proxy server forwards the 302 Server to User A Moved Temporarily message to User A. User A sends a SIP ACK to the proxy server. The ACK message notifies the ACK—User A to Proxy Server proxy server that User A has received the ACK message.
Page 452: Call Conference
User A mixes two RTP channels and therefore establishes a conference between User B and User C. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
Page 453 Appendix User A User B User C Proxy Server F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK Session1 established between User A and User B is active F9.
Page 454 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
Page 455 Appendix Step Action Description User A sends a SIP ACK to the proxy server. The ACK confirms that User A has ACK—User A to Proxy Server received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to User B.
Page 456 Administrator’s Guide for SIP-T2xP IP Phones Step Action Description the SIP INVITE request to User C. User C sends a SIP 180 Ringing response 180 Ringing—User C to Proxy to the proxy server. The 180 Ringing Server response indicates that the user is being alerted.
Page 457: Network Settings
Appendix Appendix F: Sample Configuration File This section provides the sample configuration file necessary to configure the IP phone. Any line starts with a pound sign (#) is considered to be a comment, unless the # is contained within double quotes. For Boolean fields, 0 = disabled, 1 = enabled. This file contains sample configurations for the <y0000000000xx>.cfg or <MAC>.cfg file.
Page 458: Time Settings
Administrator’s Guide for SIP-T2xP IP Phones Time Settings local_time.time_zone = local_time.time_zone_name = local_time.ntp_server1 = local_time.ntp_server2 = local_time.interval = local_time.dhcp_time = #Use the following parameters to set the time and date manually. local_time.manual_time_enable = local_time.date_format = local_time.time_format = Auto DST Settings local_time.summer_time =...
Page 459 Appendix #Hotline features.hotline_number = features.hotline_delay = #Web Server Type network.web_server_type = network.port.http = network.port.https = #DTMF Suppression features.dtmf.hide = features.dtmf.hide_delay = Call Forward # In Phone Mode features.fwd_mode = 0 forward.always.enable = forward.always.target = forward.always.on_code = forward.always.off_code = forward.busy.enable = forward.busy.target = forward.busy.on_code = forward.busy.off_code =...
Page 460 Administrator’s Guide for SIP-T2xP IP Phones account.1.timeout_fwd.off_code = Call Transfer transfer.semi_attend_tran_enable = transfer.blind_tran_on_hook_enable = transfer.on_hook_trans_enable = transfer.tran_others_after_conf_enable = Call Conference account.1.conf_type = account.1.conf_uri = DTMF account.1.dtmf.type = account.1.dtmf.dtmf_payload = account.1.dtmf.info_type = Distinctive Ring Tones account.1.alert_info_url_enable = distinctive_ring_tones.alert_info.1.text = distinctive_ring_tones.alert_info.1.ringer = Tones voice.tone.dial =...
Page 461 Appendix ldap.base = ldap.user = ldap.password = ldap.max_hits = ldap.name_attr = ldap.numb_attr = ldap.display_name = ldap.version = ldap.call_in_lookup = ldap.ldap_sort = Action URL action_url.setup_completed = action_url.log_on = action_url.log_off = action_url.register_failed = action_url.off_hook = action_url.on_hook = action_url.incoming_call = action_url.outgoing_call = action_url.call_established = action_url.dnd_on = action_url.dnd_off = action_url.always_fwd_on =...
Page 462 Administrator’s Guide for SIP-T2xP IP Phones action_url.transfer_failed = #SNMP network.snmp.enable = network.snmp.port = network.snmp.trust_ip = #Access URL of Resource Files dialplan_dialnow.url = dialplan_replace_rule.url = local_contact.data.url = remote_phonebook.data.1.url =...
Page 463 Index Index Numeric 180 Ring Workaround Call Completion 802.1x Authentication Call Forward Call Hold Call Log Call Park About This Guide Call Recording Acoustic Echo Cancellation Call Return Action URL Call Transfer Action URI Call Waiting Administrator Password Calling Line Identification Presentation Always Forward Connected Line Identification Presentation Analyzing the Configuration Files...
Page 464 Administrator’s Guide for SIP-T2xP IP Phones Early Media Missed Call Log Encrypting Configuration Files Multicast Paging Enabling the Watch Dog Feature Music on Hold Getting Information from Status Indicators NAT Traversal Getting Started Network Address Translation (NAT) Group Call Pickup...
Page 465 Index Softkey Layout Specifying the Language to Use SRTP STUN Server Suppress DTMF Display Summary of Changes Table of Contents Time and Date Transfer on Conference Hang Up Transfer via DTMF Transport Layer Security (TLS) Troubleshooting Troubleshooting Methods Troubleshooting Solutions TR-069 Device Management Upgrading Firmware Use Outbound Proxy in Dialog...

Yealink SIP-T2XP Administrator's Manual

About this Guide

Table of Contents

Product Overview

Getting Started

Configuring Basic Features

Configuring Advanced Features

Configuring Audio Features

Configuring Security Features

Upgrading Firmware

Resource Files

Troubleshooting

Appendix

Appendix A: Glossary

Appendix B: Time Zones

Appendix C: Configuration Parameters

Appendix D: SIP (Session Initiation Protocol)

Appendix E: SIP Call Flows

Quick Links

Troubleshooting

Related Manuals for Yealink SIP-T2XP

Summary of Contents for Yealink SIP-T2XP