Copyright © 2014 YEALINK NETWORK TECHNOLOGY Copyright © 2014 Yealink Network Technology CO., LTD. All rights reserved. No parts of this publication may be reproduced or transmitted in any form or by any means, electronic or mechanical, phot ocopying, recording, or otherwise, for any purpose, without the express writ ten permission of Yealink Network Technology CO., LTD. Under the law, reproducing includes translating into another language or format.
Note: This device is tested and complies with the limits for a Class B digital device, pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications.
Yealink SIP-T20P/SIP-T20 firmware contains third-party software under the GNU General Public License (GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms and conditions of the license. The original GPL license, source code of components licensed under GPL and used in Yealink products can be downloaded online: http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293.
About This Guide Thank you for choosing the SIP-T20P/SIP-T20 IP phone, exquisitely designed to provide business telephony features, such as Call Hold, Call Transfer, Busy Lamp Field, Multicast Paging and Conference over an IP network. The difference between the SIP-T20P and SIP-T20 IP phone is that the former supports PoE, while the latter does not. This guide provides everything you need to quickly use your new phone.
User Guide for the SIP-T20P/SIP-T20 IP Phone Packaging Contents on page 9 Phone Status on page 13 Documentations of the SIP-T20 IP phone have also been added.
About This Guide Major updates have occurred to the following sections: Keypad Lock on page 21 Volume on page 24 Ring Tones on page 24 Call Completion on page 59 DSS Keys on page 37 Do Not Disturb (DND) on page 61 Call Forward on page 64 Busy Lamp Field (BLF) on page 79 vii
User Guide for the SIP-T20P/SIP-T20 IP Phone viii
Table of Contents About This Guide................................................................ v In This Guide ................................................................................................................ v Summary of Changes.................................................................................................. v Changes for Release 72, Guide Version 72. 25 .......................................................... v Changes for Release 72, Guide Version 72. 1 ............
User Guide for the SIP-T20P/SIP-T20 IP Phone Key as Send ............................................................................................................ 20 Keypad Lock ........................................................................................................... 21 Audio Settings ........................................................................................................... 24 Volume.......................................................................................
Table of Contents Busy Lamp Field (BLF) ............................................................................................... 79 Call Recording ........................................................................................................... 81 Hot Desking ............................................................................................................... 82 Intercom...............................................................................................................
Overview This chapter provides the overview of the SIP-T20P/SIP-T20 IP phone. Topics include: Hardware Component Instructions Icon Instructions LED Instructions User Interfaces Documentations If you require additional information or assistance with your new phone, contact your system administrator. The main hardware components of the SIP-T20P/SIP-T20 IP phone are the LCD screen and the keypad.
User Guide for the SIP-T20P/SIP-T20 IP Phone Hardware component instructions of the SIP-T20P/SIP-T20 IP phone are: It e m De s cript ion Shows information about calls, messages, time, date and other relevant data.
Overview Icons appearing on the LCD screen are described in the following table: Icon De s cript ion Network is unavailable Hands-free speakerphone mode Handset mode Headset mode 123 Numeric input mode abc Multi-lingual lowercase letters input mode ABC Multi-lingual uppercase letters input mode 2aB Alphanumeric input mode Voice Mail Auto Answer Do Not Disturb Call Forward/Forwarded Calls Call Mute Keypad Lock Received Calls Placed Calls Missed Calls 3
User Guide for the SIP-T20P/SIP-T20 IP Phone P ower Indicator LED L ED St at us De s cript ion The phone is initializing. The phone is busy. Solid green The phone is idle. The call is placed on hold or is held. The phone receives a voice mail. Fast flashing green (300ms) Off The phone is ringing. The call is mute. The phone is powered off. L ine key LED L ED St at us Solid green De s cript ion The line is in conversation. The line is seized. Fast flashing green The line receives an incoming call.
Overview Two ways to customize configurations of your SIP-T20P/SIP-T20 IP phone: The user Interface on the IP phone. The user Interface in a web browser on your PC. The hardware components keypad and LCD screen constitute the phone user interface , which allows the user to execute all call operation tasks and basic configuration changes directly on the phone. In addition, each phone has a web user interface to access all configuration settings.
User Guide for the SIP-T20P/SIP-T20 IP Phone O pt ions P hone User Interface --Time & Date √ --Administrator Password √ --Key as Send √ --Keypad Lock √ --Ring Tones √ We b User Interface --Contact Management --Local Directory √ --Blacklist √ --Call History Management √ --Logo Customization --DSS Keys √ --Account Registration √ --Dial Plan --Emergency Number --Live Dialpad --Hotline √ Bas ic Call Features --Auto Answer √ --Auto Redial √ --Call Completion √ --Call Return √ --
Overview O pt ions P hone User Interface --Activation √ --Label √ --Display Name √ --Register Name √ --User Name √ --Password √ --SIP Server1/2 √ We b User Interface --Server Option --Registrar Port --Outbound Status √ --Outbound Proxy √ --NAT Traversal Not e --STUN Status √ --STUN Server √ The table above lists most of the feature options. Please refer to the relevant sections for more information.
User Guide for the SIP-T20P/SIP-T20 IP Phone 8
Getting Started This chapter provides basic installation instructions and information for obtaining the best performance with the SIP-T20P/SIP-T20 IP phone. Topics include: Packaging Contents Phone Installation Phone Initialization Phone Status Basic Network Settings Registration Idle Screen If you require additional information or assistance with your new phone, contact your system administrator.
User Guide for the SIP-T20P/SIP-T20 IP Phone P hone Stand Power A dapter (Optional) Handset & Handset Cord Et hernet Cable Q uick Installation Guide & Quick Reference Guide Check this list before installation. If you find anything missing, contact your system administrator.
Getting Started If your phone is already installed, proceed to Phone Initialization on page 13. This section introduces how to install the phone: 1) Attach the stand 2) Connect the handset and optional headset 3) Connect the network and power 1) A t tach t he Stand 2) Connect the Handset and optional Headset Not e A headset is not included in the packaging contents. Contact your system administrator for more information.
User Guide for the SIP-T20P/SIP-T20 IP Phone AC Power To connect the AC power: 1. Connect the DC plug on the power adapter to the DC5V port on the phone and connect the other end of the power adapter into an electrical power outlet. 2. Connect the included or a standard Ethernet cable between the Internet port on the phone and the one on the wall or switch/hub device port.
Getting Started After your phone is powered on, the system boots up and performs the following steps: A utomatic Phone Initialization The phone finishes the initialization by loading the saved configuration. The LCD screen displays ―Initializing, Please wait‖ during this process. DHCP (Dynamic Host Configuration Protocol) The phone attempts to contact a DHCP server in your network to obtain valid network settings (e.g., IP address, subnet mask, default gateway address and DNS address) by default.
User Guide for the SIP-T20P/SIP-T20 IP Phone 3. Enter the user name (admin) and password (admin) in the login page. 4. Click Conf irm to login. The phone status is displayed on the first page of the web user interface. If your phone cannot contact a DHCP server for any reason, you need to configure network settings manually. The IP phone can support either or both IPv4 and IPv6 addresses. To configure the IP address mode via phone user interface: 14 1. Press . 2.
Getting Started 3. Press or to select IP v4, IPv6 or IP v4 & IPv6 from the IP Mode field. 1. IP Mode: ◄ IPv4 & IPv6 ▶ 4. Press to accept the change. To configure a static IPv4 address via phone user interface: 1. Press . 2. Select Se ttings->Advanced (password: admin) ->Ne twork->WAN Port. 3. Press to select IP v 4 and press 4. Press to select St atic IP and press 5. Enter the desired values in the IP A ddress, Subnet Mask, Default Gateway, P ri.DNS . . and Se c.
User Guide for the SIP-T20P/SIP-T20 IP Phone To configure PPPoE via phone user interface: 1. Press . 2. Select Se ttings->Advanced (password: admin) ->Ne twork->WAN Port. 3. Press to select IP v 4 and press 4. Press to select P P PoE and press . . 1. PPPoE User: 2aB Not e 5. Enter the user name and password in the corresponding fields. 6. Press to accept the change.
Customizing Your Phone You can customize your SIP-T20P/SIP-T20 IP phone by personally configuring certain settings, for example, language, time & date and ring tones. You can also personalize different ring tones for different callers. This chapter provides basic operating instructions for customizing your phone.
User Guide for the SIP-T20P/SIP-T20 IP Phone 3. Press or to select the desired language. *1. English(English) 2. Français(French) 4. Press to accept the change. Text displayed on the LCD screen will change to the selected language. To change the language for web user interface: 1. Click on Se t tings->Preference. 2. Select the desired language from the pull-down list of L anguage. 3. Click Conf irm to accept the change.
Customizing Your Phone The default time zone is ―+8 China (Beijing)‖. 1. Time Zone: ◄ *+8 China (Beijing) ▶ 4. Enter the domain names or IP addresses in the NTP Server 1 and NTP Server 2 fields respectively. Not e 5. Press or to select the desired value from the DST field. 6. Press to accept the change. Please refer to Appendix A - Time Zones for the list of available time zones on the IP phone. To configure the time and date manually via phone user interface: 1. Press . 2.
User Guide for the SIP-T20P/SIP-T20 IP Phone and ―YY‖ represents a two-digit year which is not displayed on the LCD screen of the SIP-T20P/SIP-T20 IP phone. The date formats you need to know: Dat e F ormat Ex ample (2013-11-15) MM/DD/YY Nov 15 DD/MM/YY 15 Nov YY/MM/DD 15 Nov Time and date is configurable via web user interface at the path Settings->Time & Date. The Advanced option is only accessible to the administrator. The default administrator password is ―admin‖.
Customizing Your Phone 2. Select F e atures->Key as Send. Key as Send: ◄ *# 3. Press or ► to select # or * from the Ke y as Send field, or select Disable from the Ke y as Send field to disable this feature. 4. Press to accept the change. Key as send is configurable via web user interface at the path F e atures->General Inf ormation. You can lock the keypad of your phone temporarily when you are not using it. This feature helps you to protect your phone from unauthorized use.
User Guide for the SIP-T20P/SIP-T20 IP Phone 3. Press or to select desired type from the Ke ypad Lock field. 1. Keypad Lock: ◄ Menu Key ► 4. Press to accept the change. 5. Long press to lock the keypad immediately when the phone is idle. The LCD screen prompts ―Keypad Locked!‖ and displays the icon . Keypad Locked! You can specify the interval (in seconds) to automatically lock the keypad instead of long pressing .
Customizing Your Phone To unlock the keypad, you must know the keypad unlock PIN of the phone. The default keypad unlock PIN is ―123‖. To change the keypad unlock PIN via phone user interface: 1. Press . 2. Select Se ttings->Basic->Phone Unlock. 3. Enter the old PIN in the Current PIN field. 1. Current PIN: 123 Not e 4. Enter the new PIN in the Ne w P IN field. 5. Enter the new PIN again in the Confirm PIN field. 6. Press to accept the change. The unlock PIN length must be within 15 digits.
User Guide for the SIP-T20P/SIP-T20 IP Phone Keypad lock is configurable via web user interface at the path F e atures->Phone Lock. You can press the Volume key to adjust the ringer volume when the phone is idle. You can also press the Volume key to adjust the receiver volume of currently engaged audio devices (handset, speakerphone or headset) when the phone is in use. To adjust the volume when the phone is idle: 1. Press to adjust the ringer volume.
Customizing Your Phone 4. Press to accept the change. To s elect a ring tone for each account via web user interface: 1. Click on A ccount. 2. Select the desired account from the pull-down list of A ccount. 3. Click on Bas ic. 4. Select the desired ring tone from the pull-down list of Ring Type. If Common is selected, this account will use the ring tone selected for the phone . Refer to the above instruction. 5. Not e Click Conf irm to accept the change.
User Guide for the SIP-T20P/SIP-T20 IP Phone 2. Click Browse to locate a ring tone file (file format must be *.wav) from your local system. 3. Not e Click Upload to upload the file. All custom ring tone files must be within 100KB. Custom ring tones for your phone can be uploaded via web user interface only. This section provides the operating instructions for managing contacts.
Customizing Your Phone 5. Enter the group name in the Name field. Name: Test abc 6. Press or to select the desired group ring tone from the Ring Tones field. If A ut o is selected, this group will use the ring tone assigned to the account. For more information on ring tone for a account, refer to Ring Tones on page 24. 7. Press to accept the change. To e dit a group in the local directory: 1. Press . 2. Select L ocal Directory. 3. Press or to select the desired group (e.g., Test). 4.
User Guide for the SIP-T20P/SIP-T20 IP Phone 5. Press to accept the change. You can add contacts to your local directory in the following ways: Manually From call history Adding Contacts Manually To add a contact to the local directory manually: 1. Press . 2. Select L ocal Directory. 3. Press or to select Enter and then press . If the groups have been added to the local directory, select the desired group, then press or to select Ent er and then press 4.
Customizing Your Phone 3. Press . 4. Press to switch the input mode, and enter the contact name in the Name field. Name: abc 5. Press to accept the change. The entry is successfully saved to the local directory. To e dit a contact in the local directory: 1. Press . 2. Select L ocal Directory. 3. Press or to select Enter and then press . If the groups have been added to the local directory, select the desired group, then press 4.
User Guide for the SIP-T20P/SIP-T20 IP Phone To place a call t o a contact in the local directory: 1. Press . 2. Select L ocal Directory. 3. Press or to select Enter and then press . If the groups have been added to the local directory, select the desired group, then press or to select Ent er and then press 4. Press or to select the desired contact. 5. Press or to select Se nd. 6. Do one of the following: - .
Customizing Your Phone You can manage your phone’s local directory via phone or web user interface. But you can only import or export the contact list via web user interface. To import an XML file of contact list via web user interface: 1. Click on Directory->Local Directory. 2. Click Browse to locate a contact list file (file format must be *.xml) from your local system. 3. Click Import XML to import the contact list. The web user interface prompts ―The original contact will be covered, Continue?‖.
User Guide for the SIP-T20P/SIP-T20 IP Phone 6. (Optional.) Select the contact information you want t o import into the local direct ory from the pull down list of Index. 7. Click Import to complete importing the contact list. To e xport contact list via web user interface: 1. Click on Directory->Local Directory. 2. Click Ex port XML (or Ex port CSV). 3. Click Sav e to save the contact list to your local system. Not e Contact lists can be imported/exported via web user interface only.
Customizing Your Phone 6. Enter the name and the office, mobile or other numbers in the corresponding fields. Name: Test abc 7. Press or to select the desired account from the A ccount field. If A ut o is selected, the phone will use the first available account when placing calls to the contact from the blacklist. 8. Press to accept the change.
User Guide for the SIP-T20P/SIP-T20 IP Phone To place a call f rom the call history list: 1. Press . The LCD screen displays the call list. 2. Press or to switch between placed calls, received calls, missed calls and forwarded calls. 3. Press or to select the desired entry. 4. Press to dial out. To add a contact to the local directory from the call history list: 1. Press . The LCD screen displays the call list. 2.
Customizing Your Phone 3. Enter the desired text in the Te xt Logo field. 4. Click Conf irm to accept the change. Not e The maximum length of text logo is 15 characters. A text logo is configurable via web user interface only. Physically connect your headset and activate the headset mode for use. For more information on physically connecting a headset, refer to Phone Installation on page 11. To activate the headset: 1. Press The on the phone.
User Guide for the SIP-T20P/SIP-T20 IP Phone To deactivate the headset: 1. Press again on the phone. The headset icon disappears when the headset mode is deactivated. You can use headset in priority when enabling headset prior. This feature is especially useful for permanent or full-time headset users. To e nable Headset Prior via web user interface: 1. Click on F e atures->General Information. 2. Select Enabled from the pull-down list of He adset Prior. 3. Click Conf irm to accept the change.
Customizing Your Phone You can use two headsets when enabling dual headset. To use this feature, you must physically connect headsets to the headset jack and handset jack respectively. Once the phone connects to a call, the headset connected to the headset jack will have full-duplex capabilities, while the one connected to the handset jack will only be able to listen. To e nable Dual Headset via web user interface: Not e 1. Click on F e atures->General Information. 2.
User Guide for the SIP-T20P/SIP-T20 IP Phone You can assign predefined functionalities to the line keys located on the right of the phone. Line keys allow you to have quick access to features such as call return or voice mail. The line key LEDs will indicate status when the keys are assigned with particular features, such as BLF. The default key type of each line key is Line. To configure the line key via phone user interface: 1. Press . 2. Select F e atures->DSS Keys. 3.
Customizing Your Phone Speed Dial You can use this key feature to speed up dialing the numbers often used or hard to remember. De pendencies: Type (Speed Dial) Account ID (the account this feature will be applied to) Value (the number you want to dial out) Us age: Press the DSS key to dial out the number specified in the Value field, using the account selected from the A ccount ID field. Voice Mail You can use this key feature to connect voice mail quickly.
User Guide for the SIP-T20P/SIP-T20 IP Phone L ocal Group You can use this key feature to access the contact group in the local directory quickly. For more information, refer to Local Directory on page 26. De pendencies: Type (Key Event) Ke y Type (Local Group) Local Group (the contact group you want to access) Usage: Press the DSS key to access the contact group specified in the Local Group field. Conference You can use this key feature to set up a conference call.
Customizing Your Phone Us age: When the transfer mode on DSS key is Blind Transfer, press the DSS key to complete the blind transfer to the number specified in the Value field. When the transfer mode on DSS key is A t tended Transfer, press the DSS key to dial out the number specified in the Value field, and then you can perform the attended or semi-attended transfer.
User Guide for the SIP-T20P/SIP-T20 IP Phone G roup Listening You can use this key feature to activate the Speakerphone and Handset/Headset mode at the same time. It is suitable for the group conversation which has more than one person at one side. You are able to speak and listen through the handset/headset, meanwhile the others nearby can only listen through the speaker. De pendencies: Type (Key Event) Ke y Type (Group Listening) Us age: 1.
Customizing Your Phone 2. Customize specific features for these keys. 3. Click Conf irm to accept the change. You can click Re s et to default to reset custom settings to defaults. Not e Programmable keys are configurable via web user interface only. You can register two accounts at most on the SIP-T20P/SIP-T20 IP phone. You can also configure each line key to associate with an account or configure two line keys to associate with an account. To register an account via phone user interface: 1.
User Guide for the SIP-T20P/SIP-T20 IP Phone To disable an account via phone user interface: 1. Press . 2. Select Se ttings->Advanced ->A ccounts. 3. Select the desired account. 4. Select Disable from the A ct ivation field. 5. Press to accept the change. Account registration is configurable via web user interface at the path A ccount->Register. You can configure two line keys to associate with an account. This enhances call visualization and simplifies call handling.
Customizing Your Phone ―91([5-7])1(x)‖ would match ―915 11‖, ―916 18‖, ―917 15‖. The ―$‖ should be followed by the sequence number of a parenthesis. The ―$‖ plus the sequence number means the whole character or characters placed in the parenthesis. The number directs to the right parenthesis when there are $ more than one. Example: A replace rule configuration: Prefix: "9([5-7]) (.)", Replace: "5$2".
User Guide for the SIP-T20P/SIP-T20 IP Phone 5. Click A dd to add the replace rule. When you enter the number ―1‖ using the keypad and then press the OK key, the phone will dial out ―1234‖ instead. Not e The valid values of Account field can be one or more digits between 1 and 2. Each digit must be separated by a comma. For example, when you enter the value ―1, 2‖ in the Account field, this replace rule will apply to account1 and account2.
Customizing Your Phone 3. Enter the desired line ID in the A ccount field or leave it blank. For more information on the valid values of A ccount field, refer to Replace Rule on page 45. 4. Click A dd to add the dial-now rule. When you enter the number ―1234‖ using the keypad, the phone will dial out ―1234‖ automatically without pressing any key. Not e You can also edit or delete the dial-now rule, refer to Replace Rule on page 45 for more information.
User Guide for the SIP-T20P/SIP-T20 IP Phone 2. Enter the desired time within 1-14 (in seconds) in the Time-Out for Dial-Now Rule field. 3. Not e Click Conf irm to accept the change. Delay time for dial-now rule is configurable via web user interface only. Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in a count ry. This feature is necessary only when dialing the number outside the code area.
Customizing Your Phone 3. Enter the desired line ID in the A ccount field or leave it blank. For more information on the valid values of A ccount field, refer to Replace Rule on page 45. 4. Not e Click Conf irm to accept the change. The default values of minimum and maximum lengths are 1 and 15 respectively. Area code is configurable via web user interface only. You can block specific numbers from being dialed on your phone.
User Guide for the SIP-T20P/SIP-T20 IP Phone 3. Enter the desired line ID in the A ccount field or leave it blank. For more information on the valid values of A ccount field, refer to Replace Rule on page 45. 4. Not e Click Conf irm to add the block out number. Block out number is configurable via web user interface only.
Customizing Your Phone 2. Enter the emergency number in the Emergency field. For multiple numbers, enter a comma between every two emergency numbers. The default emergency numbers are 112, 911 and 110. 3. Not e Click Conf irm to accept the change. Emergency number is configurable via web user interface only. You can enable live dialpad on the SIP-T20P/SIP-T20 IP phone, which enables the IP phone to automatically dial out phone numbers without pressing any other key.
User Guide for the SIP-T20P/SIP-T20 IP Phone 3. Enter the desired delay time in the Int er Digit Time (1~14s) field. The default delay time is 4s. 4. Not e Click Conf irm to accept the change. Live dialpad is configurable via web user interface only. You can dial a hotline number immediately upon lifting the handset, pressing the Speakerphone key or the line key. You can also configure a delay, where the phone will dial out the hotline number automatically after the specified period of time.
Basic Call Features The SIP-T20P/SIP-T20 IP phone is designed to be easily used like a regular phone on a public switched telephone network (PSTN). You can place calls, answer calls, transfer a call to someone else, or conduct a conference call. This chapter provides basic operating instructions for the SIP-T20P/SIP-T20 IP phone.
User Guide for the SIP-T20P/SIP-T20 IP Phone Using the headset You can also dial the number first, and then choose the way you want to speak to the other party. You can also search and dial a contact from call history or local directory. For more information, refer to Contact Management on page 26 and Call History Management on page 33. During a call, y ou can alternate between Speakerphone, Headset, and Handset modes by pressing the Speakerphone key, HEADSET key, or picking up the handset.
Basic Call Features To place a call using t he headset: Do one of the following: With the optional headset connected, press - to activate headset mode. Press the line key to obtain a dial tone. Enter the desired number using the keypad. Press or . With the optional headset connected, press - to activate headset mode. Enter the desired number using the keypad. Press Not e or . To permanently activate the headset mode, refer to Headset Prior on page 36.
User Guide for the SIP-T20P/SIP-T20 IP Phone To answer a call using the handset: 1. Pick up the handset. To answer a call using the hands-free speakerphone mode: Do one of the following: - Press . - With the handset on-hook and headset mode deactivated, press - With the handset on-hook and headset mode deactivated, press the line key (the . line LED flashes green). To answer a call using the headset: Do one of the following: - Press .
Basic Call Features To redial the last dialed number from your phone: 1. Press twice. A call to your last dialed number is attempted. To redial a previously dialed number from your phone: 1. Press 2. Press press when the phone is idle. or to select the desired entry from the placed calls list, and then or . You can use auto answer to automatically answer an incoming call on a line. Auto answer is configurable on a per-line basis. To configure auto answer via phone user interface: 1. Press .
User Guide for the SIP-T20P/SIP-T20 IP Phone You can enable auto redial to redial the phone number automatically when the called party is busy. You can also configure the times settings for auto redial and the time to wait between redial attempts. To configure auto redial via phone user interface: 1. Press . 2. Select F e atures->Auto Redial. 3. Press or to select Enable from the A ut o Redial field. 1. Auto Redial: ◄ Enable 4. ► Enter the desired time in the Interval field.
Basic Call Features You can use call completion to notify the caller who failed to reach a desired callee when the callee becomes available to receive a call. To configure call completion via phone user interface: 1. Press . 2. Select F e atures->Call Completion. 1. Call Completion: ◄ Enable ► 3. Press or to select Enable from the Call Completion field. 4. Press to accept the change. Call completion is configurable via web user interface at the path F eatures->General Inf ormation.
User Guide for the SIP-T20P/SIP-T20 IP Phone You can press a call return key to place a call back to the last incoming call. To configure a call return key via phone user interface: 1. Press . 2. Select F e atures->DSS Keys. 3. Select a desired DSS key. 4. Press or to select Ke y Event from the Type field. 5. Press or to select Call Return from the Ke y Type field. 2. Key Type: ◄ Call Return 6. Press ► to accept the change.
Basic Call Features To place a call on hold: 1. Press during a call. The LCD screen indicates the call is on hold and the line LED flashes green. 1234 Hold Not e 01:20 The phone will beep softly every 30 seconds to remind you that you still have a call on hold. To resume a held call: 1. Press again. Multiple Calls on Hold: If multiple calls are placed on hold, do one of the following: - Press or to switch between the calls, and then press to retrieve the desired call.
User Guide for the SIP-T20P/SIP-T20 IP Phone Not e 2. In the DND block, mark the desired radio box in the Mode field. 3. Click Conf irm to accept the change. DND mode is configurable via web user interface only. To activate DND in phone mode: 1. Press . 2. Select F e atures->DND Code. 3. Press 4. Select Enable from the DND field. or to select P hone Mode from the DND Account field. 2. DND: ◄ Enable 5. ► (Optional.
Basic Call Features Incoming calls will be rejected automatically and "n Missed Call" ("n" indicates the number of the missed calls) will appear on the LCD screen. 1 Missed Call 18 Jan 20:35 To activate DND in custom mode: 1. Press . 2. Select F e atures->DND Code. 3. Press 4. Select Enable from the DND field. or to select the desired account from the DND Account field. 2. DND: ◄ Enable 5. ► (Optional.
User Guide for the SIP-T20P/SIP-T20 IP Phone To configure the DND authorized numbers via web user interface: 1. Click on F e atures->Forward & DND. 2. Select Enabled from the pull-down list of DND Emergency. 3. Enter the numbers in the DND Authorized Numbers field. For multiple numbers, enter a comma between every two numbers. 4. Click Conf irm to accept the change.
Basic Call Features No A nswer Forward: Incoming calls are forwarded if not answered after a period of time. You can enable/disable call forward for the phone system, or you can customize call forward for each or all accounts. Two call forward modes: P hone (default): Call forward is effective for the phone system. Custom: Call forward can be configured for each account. To configure the call forward mode via web user interface: Not e 1. Click on F e atures->Forward & DND. 2.
User Guide for the SIP-T20P/SIP-T20 IP Phone the F orward To field. 3) (Optional.) Enter the always forward on code and off code respectively in the O n Code and O f f Code field. 1. Always: ◄ Enable b.) ► If you select Bus y Forward: 1) Press or to select Enable from the Busy field. 2) Enter the destination number you want to forward incoming calls to when the phone is busy in the F orward To field. 3) (Optional.
Basic Call Features 4. Press 5. Depending on your selection: a.) or to select the forwarding type, and then press . If you select A lways Forward: 1) Press or to select Enable from the A lways field. 2) Enter the destination number you want to forward all incoming calls to in the F orward To field. 3) (Optional.) Enter the always forward on code and off code respectively in the O n Code and O ff Code field. 1. Always: ◄ Enable b.
User Guide for the SIP-T20P/SIP-T20 IP Phone The icon appears on the LCD screen and the prompt ―F W D:‖ appears in front of the associate account. FWD: 1003 24 Jul 10:39 Call forward is configurable via web user interface at the path F e atures->Forward & DND . Not e You can also enter the SIP URL or IP address in the Fo rward To field. For more information on using the SIP URL or IP address, refer to Placing Calls on page 53.
Basic Call Features 2. Enter the number you want to forward the incoming call to. FWD: 1234 123 3. Press or . The LCD screen prompts a call forwarded message. FWD To 1234 12 Sep 10:53 You can transfer a call to another party in one of the three ways: Blind Transfer: Transfer a call directly to another party without consulting. Se mi-Attended Transfer: Transfer a call when the target phone is ringing. A t tended Transfer: Transfer a call with prior consulting.
User Guide for the SIP-T20P/SIP-T20 IP Phone 2. Enter the number you want to transfer the call to. 3. Press 4. After the party answers the call, press or to dial out. to complete the transfer. If you are using a handset, the transfer can be completed by hanging up the handset. You can cancel the transfer before the call is connected by pressing . You can enable or disable call waiting on the phone. If call waiting is enabled, you can receive another call when there is an active call on the phone.
Basic Call Features The SIP-T20P/SIP-T20 IP phone supports up to 3 parties (including yourself) in a conference call. This is the default method of conference called Local Conference. To s et up a local conference call: 1. Place a call to the first party. 2. When the first party answers the call, press to place a new call. The active call is placed on hold. 3. Enter the number of second party and press or . Dial: 1234 123 4.
User Guide for the SIP-T20P/SIP-T20 IP Phone 2. Select the desired account from the pull-down list of A ccount. 3. Click on A dv anced. 4. Select Ne t work Conference from the pull-down list of Conference Type. 5. Enter the conference URI (e.g., conference@example.com) in the Conference URI field. 6. Not e Click Conf irm to accept the change. Network conference is configurable via web user interface only. To s et up a network conference call: 1. Place a call to the first party. 2.
Basic Call Features The procedures to set up a network conference call for specific servers may be different from that introduced above. Contact your system administrator for more information. You can use call park to place a call on hold, and then retrieve the call from another phone in the system (for example, a phone in another office or conference room). You can park the active call by pressing the call park key on the phone.
User Guide for the SIP-T20P/SIP-T20 IP Phone The system establishes call between phone C and B. Not e The call park code and call park retrieve code are predefined on the system server. Contact your system administrator for more information. If the parked call is not retrieved within a period of time assigned by the system, the phone performing call park feature will receive call back. You can use call pickup to answer someone else’s incoming call on the phone.
Basic Call Features To pick up a call directly: 1. Press the pickup key on your phone when the specific phone number receives an incoming call. The incoming call is answered on your phone. Group Call Pickup To configure a group pickup key via phone user interface: 1. Press . 2. Select F e atures->DSS Keys. 3. Select the desired DSS key. 4. Press or to select Ke y Event from the Ty pe field. 5. Press or to select G roup Pickup from the Ke y Type field. 6.
User Guide for the SIP-T20P/SIP-T20 IP Phone To configure anonymous call via phone user interface: 1. Press . 2. Select F e atures->Anonymous Call. 3. Press or to select the desired line from the L ine ID field. 4. Press or to select Enable from the Se nd Anon field. 2. Send Anon: ◄ Enable or ► 5. (Optional.) Press to select the desired value from the A non Code field. 6. (Optional.) Enter the anonymous call on code in the Call On Code field. 7. (Optional.
Basic Call Features 5. Press or to select Enable from the A non Rejection field. 5. Anon Rejection: ◄ Enable ► 6. (Optional.) Enter the anonymous call rejection on code in the Re ject O n Code field. 7. (Optional.) Enter the anonymous call rejection off code in the Re ject Off Code field. 8. Press to accept the change. Anonymous call rejection is configurable via web user interface at the path A ccount->Basic.
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Advanced Phone Features This chapter provides operating instructions for the advanced features of the SIP-T20P/SIP-T20 IP phone. Topics include: Busy Lamp Field (BLF) Call Recording Hot Desking Intercom Multicast Paging Music on Hold Automatic Call Distribution (ACD) Message If you require additional information or assistance with your new phone, cont act your system administrator. You can use BLF to monitor a specific line for status changes on the phone.
User Guide for the SIP-T20P/SIP-T20 IP Phone 6. Enter the phone number you want to monitor in the Value field. 7. (Optional.) Enter the pickup code in the Ex tension field. 8. Press to accept the change. BLF key is configurable via web user interface at the path DSSKey->Line Key. You can enable audio alert for BLF pickup on the phone. This allows the monitoring phone to play a warning tone when the monitored line receives an incoming call. To e nable audio alert via web user interface: 1.
Advanced Phone Features You can record calls by pressing a record key on the phone. The SIP-T20P/SIP-T20 IP phone supports record and URL record. Two ways of call recording: Re cord: the phone sends SIP INFO message containing a specific header ―Record: on/off ‖ to trigger a recording. URL Record: the phone sends HTTP URL request to trigger a recording. Contact your system administrator for the predefined URL. Not e Call record is not available on all servers.
User Guide for the SIP-T20P/SIP-T20 IP Phone D SSKey->Line Key. The record and URL record keys control recording, and are available: During an active call When calls are on hold or mute During a blind or attended transfer During a conference call When the phone prompts you to answer an incoming call The record and URL record key is not available when: There are no connected calls on your phone You place a new call To record a call: 1.
Advanced Phone Features To configure a hot desking key v ia phone user interface: 1. Press . 2. Select F e atures->DSS Keys. 3. Select the desired DSS key. 4. Press or to select Ke y Event from the Ty pe field. 5. Press or to select Hot Desking from the Ke y Type field. 2. Key Type: ◄ Hot Desking 6. Press ► to accept the change. Hot desking key is configurable via web user interface at the path DSSKey->Line Key. To use hot desking: 1. Press the hot desking key when the phone is idle.
User Guide for the SIP-T20P/SIP-T20 IP Phone Intercom is a useful feature in an office environment to quickly connect with the operator or the secretary. You can press the intercom key to automatically connect with a remote extension for outgoing intercom calls, and the remote extension will automatically answer the incoming intercom calls. Not e Intercom is not available on all servers. Contact your system administrator for more information. To configure an intercom key via phone user interface: 1.
Advanced Phone Features Intercom features you need to know: Int ercom Feature De s cript ion Enable or disable the IP phone to automatically Accept Intercom answer an incoming intercom call. Enable or disable the microphone on the IP phone for Intercom Mute intercom calls. Enable or disable the IP phone to play a warning Intercom Tone tone when it receives an incoming intercom call.
User Guide for the SIP-T20P/SIP-T20 IP Phone Int ercom Tone You can enable or disable the phone to play a warning tone when receiving an intercom call. If Intercom Tone is enabled, the phone plays a warning tone before answering the intercom call. If Intercom Tone is disabled, the phone automatically answers the intercom call without warning. Intercom Tone is enabled by default.
Advanced Phone Features Multicast paging key is configurable via web user interface at the path D SSKey->Line Ke y. You can also configure the phone to use a default codec for sending multicast RTP stream via web user interface. To configure a default codec for multicast paging: 1. Click on F e atures->General Information. 2. Select the desired codec from the pull-down list of Multicast Codec. The default codec is G722. 3. Not e Click Conf irm to accept the change.
User Guide for the SIP-T20P/SIP-T20 IP Phone The following figure shows a multicast RTP session on the phone: 224.5.6.20:10008 HD 00:35 2. Press to place the current multicast RTP session on hold. 3. Press to cancel the multicast RTP session. Not e Multicast RTP is one way only- from sender to the multicast address(es) (receiver). For outgoing RTP multicasts, all other existing calls on the phone will be placed on hold.
Advanced Phone Features 3. Select the desired value from the pull-down list of Paging Priority A ctive. 4. Enter the multicast IP address(es) and port number(e.g., 224.5.6.20:10008) which the phone listens for incoming RTP multicast in the L istening Address field. The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255. 5. Enter the label in the L abel field. Label will appear on the LCD screen when receiving the RTP multicast. 6. Not e Click Conf irm to accept the change.
User Guide for the SIP-T20P/SIP-T20 IP Phone 3. Click on A dv anced. 4. Enter the SIP URI (e.g., sip:moh@sip.com) in the Mus ic Server URI field. 5. Click Conf irm to accept the change. When you placed a call on hold, the held party can hear the music. Not e All involved parties cannot use encrypted RTP. Music on hold server is configurable via web user interface only. ACD is often used in offices for customer service, such as call center.
Advanced Phone Features 4. Press or to select A CD from the Ty pe field. 1. Type: ◄ ACD 5. Press ► to accept the change. ACD key is configurable via web user interface at the path DSSKey->Line Key. To log into the ACD system: 1. Press the ACD key when the phone is idle. The LCD screen prompts you the following information: Us er ID: the identify used to log into the queue. Password: the password used to log into the queue. 1. User ID: 123 2. Not e Press to login.
User Guide for the SIP-T20P/SIP-T20 IP Phone You can leave voice mails for someone else on the SIP-T20P/SIP-T20 IP phone. You can also listen to the voice mails stored in a centralized location. When receiving a new voice mail, the phone will play a warning t one and the MESSAGE key LED will illuminate and the LCD screen will display a flashing icon. New Voice Mail 27 Sep Not e 11:35 The voice mail feature is not available on all servers. Contact your system administrator for more information.
Advanced Phone Features or 2. Not e to dial out the voice mail access code. Follow the voice prompt to listen to voice mails. Before listening to the voice mails, make sure the voice mail access code has been configured. When all new voice mails are retrieved, the MESSAGE key LED will go out. To v iew voice mails via phone user interface: 1. Press . 2. Select Me ssages->View Voice Mail. The LCD screen displays the amount of new and old voice mails. 1. 1234 2 new 1 old Mail 3.
User Guide for the SIP-T20P/SIP-T20 IP Phone O pt ion De s cript ion subscribe for MWI and the voice mail number in advance. Not e Whether the phone sends SUBSCRIBE messages for MWI service to the account or the voice mail number depends on the server. Contact your system administrator for more information. To e nable MWI subscription via web user interface: 1. Click on A ccount. 2. Select the desired account from the pull-down list of A ccount. 3. Click on A dv anced. 4.
Advanced Phone Features 5. Enter the desired voice mail number in the Voice Mail field. 6. Click Conf irm to accept the change. The IP phone will subscribe to the voice mail number for MWI service using Subscribe MWI to Voice Mail. Not e MWI subscription is configurable via web user interface only.
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Troubleshooting This chapter provides general troubleshooting information t o help to solve the problems you might encounter when using your SIP-T20P/SIP-T20 IP phone. If you require additional information or assistance with your new phone, contact your system administrator. W hy is t he LCD screen blank? Ensure that the phone is properly plugged into a functional AC outlet. Ensure that the phone is plugged into a socket controlled by a switch that is on.
User Guide for the SIP-T20P/SIP-T20 IP Phone How t o obtain the MAC address of a phone when the phone is not powered on? Three ways to obtain the MAC address of a phone: You can ask your supplier for shipping information sheet which includes MAC addresses according to the corresponding PO (Purchase Order). You can find the MAC address in the label of carton box. You can also find the MAC address from the phone’s bar code on the back of the phone.
Troubleshooting Activation/Deactivation on page 35. Check that the headset volume is adjusted to an appropriate level. Refer to Volume on page 24. W hat will happen if I connect both PoE cable and power adapter? Which has t he priority? The phones manufactured before February 2010 use the power adapter preferentially, otherwise the phones use PoE preferentially. W hat is the difference between user name, register name and display name? Both user name and register name are defined by the server.
User Guide for the SIP-T20P/SIP-T20 IP Phone 3. Enter the desired time (in seconds) in the P lay Hold Tone Delay field. 4. Click Conf irm to accept the change. How t o change the user password? To change the user password via web user interface: Not e 1. Click on Se curity->Password. 2. Select us er from the pull-down list of Us er Type. 3. Enter the new user password in the Ne w Password and Confirm Password fields. 4. Click Conf irm to accept the change.
Troubleshooting How t o make a call using SRTP? You can enable SRTP to encrypt the audio stream(s) of phone calls. The parties participating in the call should enable SRTP. You can enable SRTP on a per-line basis. To e nable SRTP via web user interface: Not e 1. Click on A ccount. 2. Select the desired account from the pull-down list of A ccount. 3. Click on A dv anced. 4. Select the desired value from the pull-down list of RTP Encryption (SRTP). 5. Click Conf irm to accept the change.
User Guide for the SIP-T20P/SIP-T20 IP Phone 2. Not e Click Re boot to reboot the phone. Any reboot of your phone may take a few minutes. How t o export PCAP trace? We may need you to provide a PCAP trace to help analyze your problem. To e xport a PCAP trace via web user interface: 1. Click on Se t tings-> Configuration. 2. Click St art to begin recording signal traffic. 3. Recreate the error to be documented in the trace. 4. Click St op to end recording. 5.
Troubleshooting How t o export system log? We may need you to provide a system log to help analyze your problem. To e xport a system log via web user interface: 1. Click on Se t tings->Configuration. 2. Select 6 from the pull-down list of Sy s tem Log Level. 3. Click Conf irm to accept the change. The web user interface prompts ―Do you want to restart your machine?‖. The configuration will take effect after reboot. 4. Click O K to reboot the phone.
User Guide for the SIP-T20P/SIP-T20 IP Phone 2. Click Ex port to open file download window, and then save the file to your local system. To import the phone configurations via web user interface: 1. Click on Se t tings->Configuration. 2. Click Brower to locate a configuration file from your local system. 3. Click Import to import the configuration file. Not e The file format of configuration file must be *.bin. How t o upgrade firmware? To upgrade firmware via web user interface: 1.
Troubleshooting 4. Click O K to confirm upgrading. How t o reset t he phone? Reset the phone when other troubleshooting suggestions do not correct the problem. You need to note that all customized settings will be overwritten after resetting. So we recommend asking your system administrator for advice before resetting the phone. To reset the phone via phone user interface: 1. Press . 2. Select Se ttings->Advanced (password: admin). 3. Press or to scroll to Re s et Factory, and then press .
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Regulatory Notices Contact your Yealink Authorized Reseller for information about service agreements applicable to your product.
User Guide for the SIP-T20P/SIP-T20 IP Phone Env ironmental Requirements Place the device at a well-ventilated place. Do not expose the device under direct sunlight. Keep the device dry and free of dusts. Place the device on a stable and level platform. Please place no heavy objects on the device in case of damage and deformation caused by the heavy load. Keep at least 10 cm between the device and the closest object for heat dissipation.
Regulatory Notices Cle aning Requirements Before cleaning the device, stop using it and disconnect it from the power supply. Use a piece of soft, dry and anti-static cloth to clean the device. Keep the power plug clean and dry. Using a dirty or wet power plug may lead to electric shock or other perils.
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Appendix A – Time Zones Time Zone −11:00 −10:00 −10:00 −09:00 −08:00 −08:00 −08:00 −07:00 −07:00 −07:00 −07:00 −06:00 −06:00 −06:00 −06:00 −05:00 −05:00 −05:00 −05:00 −04:30 −04:00 −04:00 −04:00 −04:00 −04:00 −04:00 −03:30 −03:00 −03:00 −03:00 −03:00 −02:00 −01:00 0 0 0 0 0 0 0 0 +01:00 +01:00 +01:00 +01:00 +01:00 +01:00 +01:00 +01:00 +01:00 +01:00 Time Zone Name Samoa United States-Hawaii-Aleutian United States-Alaska-Aleutian United States-Alaska Time Canada(Vancouver, Whitehorse) Mexico(Tijuana, Mexica
User Guide for the SIP-T20P/SIP-T20 IP Phone Time Zone +01:00 +01:00 +01:00 +01:00 +01:00 +01:00 +01:00 +02:00 +02:00 +02:00 +02:00 +02:00 +02:00 +02:00 +02:00 +02:00 +02:00 +02:00 +02:00 +02:00 +02:00 +03:00 +03:00 +03:00 +03:30 +04:00 +04:00 +04:00 +04:00 +04:00 +04:30 +05:00 +05:00 +05:00 +05:00 +05:30 +06:00 +06:00 +07:00 +07:00 +08:00 +08:00 +08:00 +09:00 +09:00 +09:30 +09:30 +10:00 +10:00 +10:00 +10:00 +10:30 +11:00 +12:00 +12:45 +13:00 112 Time Zone Name Germany(Berlin) Hungary(Budapest) Italy(Rome
Index A About This Guide v Account Management 43 Account Registration 43 Adding Contacts 28 Adding Groups 26 Administrator password 20 Always Forward 64 Anonymous Call 75 Anonymous Call Rejection 76 Answering Calls 55 Area Code 48 Attaching Stand 11 Attended Transfer 69 Audio Settings 24 Automatic Call Distribution (ACD) B Basic Network Settings Blacklist 32 Blind Transfer 69 Block Out 49 Busy Forward 64 Busy Lamp Field (BLF) 14 79 C Call Completion 59 Call Forward 64 Call History Management 33 Call Hol
User Guide for the SIP-T20P/SIP-T20 IP Phone Network Conference Network Connection No answer Forward 71 11 64 O Outgoing Intercom Call Overview 1 84 P Packaging Contents 9 Phone Installation 11 Phone Initialization 13 Phone Status 13 Phone User Interface 5 Placing Calls 53 Placing Calls to Contacts 30 Programmable Keys 42 R Receiving RTP Stream 88 Redialing Numbers 57 Replace Rule 45 Ring Tones 24 S Safety Instructions 107 Searching for Contacts 30 Semi-attended Transfer 69 Sending RTP Stream 86 Servic
