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

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Hello, I'm simply looking for details on how to change the names of the people on my direct line userface. Do i need to sign in via my I.P Address to access this?

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Summary of Contents for Yealink SIP-T4X

  • Page 2 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 Do not dispose of WEEE as unsorted municipal waste and have to collect such WEEE separately. We are striving to improve our documentation quality and we appreciate your feedback. Email your opinions and comments to DocsFeedback@yealink.com.
  • Page 4 Yealink SIP-T4X IP phones firmware contain third-party software under the GNU General Public License (GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms and conditions of the license. The original GPL license, source code of components licensed under GPL and used in Yealink products can be downloaded online: http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293.

  • Page 5: About This Guide

     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. The information detailed in this guide is applicable to the firmware version 71 or higher.
  • Page 6 Administrator’s Guide for SIP-T4X IP Phones Chapter 3, “Configuring Basic Features” describes how to configure basic features  on IP phones. Chapter 4, “Configuring Advanced Features” describes how to configure  advanced features on IP phones. Chapter 5, “Configuring Audio Features”...
  • Page 7 About This Guide Auto Answer on page  SNMP on page  Audio Codecs on page  Encrypting Configuration Files on page  This version is updated to incorporate T41P as one of the T4X device models. The following section is new for this version: Logo Customization on page ...
  • Page 8 Administrator’s Guide for SIP-T4X IP Phones Upgrading Firmware on page  Configuring DSS Key on page  viii...

  • Page 9: Table Of Contents

    Product Overview ..............1 VoIP Principle ............................ 1 SIP Components..........................2 SIP IP Phone Models ........................3 Physical Features of SIP-T4X IP Phones ................... 4 Key Features of SIP-T4X IP Phones ................... 6 Getting Started ..............9 Connecting the IP Phone ......................... 9 Initialization Process Overview ....................
  • Page 10 Administrator’s Guide for SIP-T4X IP Phones Block Out ..........................31 Configuring Basic Features ..........33 Wallpaper ............................34 Backlight ............................36 User Password..........................38 Administrator Password ........................ 39 Phone Lock ............................. 40 Time and Date ..........................42 Language ............................47 Loading Language Packs ......................

  • Page 11: Table Of Contents

    Table of Contents Web Server Type.......................... 104 Calling Line Identification Presentation ..................106 Connected Line Identification Presentation ................107 DTMF ............................. 108 Suppress DTMF Display ......................111 Transfer via DTMF ........................112 Intercom............................113 Outgoing Intercom Calls ...................... 113 Incoming Intercom Calls ...................... 114 Configuring Advanced Features ........
  • Page 12 Administrator’s Guide for SIP-T4X IP Phones Voice Activity Detection ....................... 190 Comfort Noise Generation ....................191 Jitter Buffer ..........................192 Configuring Security Features ..........195 Transport Layer Security ......................195 Secure Real-Time Transport Protocol ..................201 Encrypting Configuration Files ....................203 Upgrading Firmware ............
  • Page 13 Table of Contents What do “on code” and “off code” mean?................ 227 How to solve the IP conflict problem? ................228 How to reset your phone to factory configurations? ............228 How to restore the administrator password? ..............228 Appendix ................229 Appendix A: Glossary .........................
  • Page 14 Administrator’s Guide for SIP-T4X IP Phones...

  • Page 15: Product Overview

    Product Overview This chapter contains the following information about SIP-T4X IP phones: VoIP Principle  SIP Components  SIP IP Phone Models  VoIP VoIP (Voice over Internet Protocol) is a technology using the Internet Protocol instead of traditional Public Switch Telephone Network (PSTN) technology for voice communications.
  • Page 16 Administrator’s Guide for SIP-T4X IP Phones SIP provides capabilities to: Determine the location of the target endpoint -- SIP supports address resolution,  name mapping, and call redirection. Determine the media capabilities of the target endpoint -- Via Session Description ...
  • Page 17 SIP-T41P  SIP-T4X IP phones comply with the SIP standard (RFC 3261), and they can only be used within a network that supports this type of phone. For successfully operating as SIP endpoints in your network, SIP-T4X IP phones must meet the following requirements: A working IP network is established.

  • Page 18: Physical Features Of Sip-T4X Ip Phones

    Administrator’s Guide for SIP-T4X IP Phones A call server is active and configured to receive and send SIP messages.  This section lists the available physical features of SIP-T4X IP phones. SIP-T46G Physical Features: 4.3” TFT-LCD, 480 x 272 pixel, 16.7M colors...
  • Page 19 Product Overview SIP-T42G Physical Features: 192 x 64 graphic LCD 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated HD Voice: HD Codec, HD Handset, HD Speaker 35 keys including 6 line keys 1xRJ9 (4P4C) handset port 1xRJ9 (4P4C) headset port 2xRJ45 10/100/1000M Ethernet ports 1XRJ12 (6P6C) EHS36 headset adapter port 10 LEDs: 1xpower, 6xline, 1xmute, 1xheadset, 1xspeakerphone Power adapter: AC 100~240V input and DC 5V/1.2A output...
  • Page 20 10 LEDs: 1xpower, 6xline, 1xmute, 1xheadset, 1xspeakerphone Power adapter: AC 100~240V input and DC 5V/1.2A output Power over Ethernet (IEEE 802.3af) In addition to physical features introduced above, SIP-T4X IP phones also support the following key features when running the latest firmware: Phone Features ...
  • Page 21 Product Overview caller identity, auto answer. Advanced Features: BLF, server redundancy, distinctive ring tones, remote phone book, LDAP , 802.1x authentication. Codecs and Voice Features  Wideband codec: G.722 Narrowband codec: G.711, G.723.1, G.726, G.729AB, GSM, iLBC . VAD, CNG, AEC, PLC, AJB, AGC Full-duplex speakerphone with AEC Network Features ...
  • Page 22 Administrator’s Guide for SIP-T4X IP Phones...

  • Page 23: Getting Started

     Configuring Basic Network Parameters  Creating Dial Plan  This section introduces how to install SIP-T4X IP phones with the components in packaging contents. Attach the stand Connect the handset and optional headset Connect the network and power Note A headset, wall mount bracket and power adapter are not included in packaging contents.
  • Page 24 Yealink Wall Mount Note For more information on how to mount the phone to a wall, refer to Quick Installation Guide for SIP-T4X IP Phones Connect the handset, optional headset and Bluetooth headset: Note Wireless headset adapter EHS36 and Bluetooth USB dongle should be purchased separately.
  • Page 25 Getting Started Connect the included or a standard Ethernet cable between the Internet port on IP phones and the one on the wall or switch/hub device port. Power over Ethernet With the included or a regular Ethernet cable, IP phones can be powered from a PoE-compliant switch or hub.
  • Page 26 19. Contacting the auto provisioning server SIP-T4X IP phones support the FTP , TFTP , HTTP , and HTTPS protocols for auto provisioning and are configured by default to use TFTP protocol. If IP phones are configured to obtain configurations from the TFTP server, they will connect to the TFTP server and download the configuration file(s) during booting up.

  • Page 27: Phone User Interface

    Getting Started Downloading the resource files In addition to configuration file(s), IP phones may require resource files before it can deliver service. These resource files are optional, but if some particular features are being deployed, these files are required. The followings show examples of resource files: Language packs ...
  • Page 28 CFG file is named as the MAC address of IP phones. For example, if the MAC address of a SIP-T46G IP phone is 001565113af5, the names of these two configuration files must be: y000000000028.cfg and 001565113af5.cfg. The name of the Common CFG file for each SIP-T4X IP phone model is: SIP-T46G: y000000000028.cfg ...
  • Page 29  The latest value configured on the IP phone takes effect finally. Icons associated with different features may appear on the LCD screen. The following table provides a description for each icon on SIP-T4X IP phone models. T46G T42G/T41P Description...
  • Page 30 Administrator’s Guide for SIP-T4X IP Phones T46G T42G/T41P Description Call Forward Call Hold Call Mute Ringer volume is 0 Phone Lock Multi-lingual lowercase letters input mode Multi-lingual uppercase letters input mode Alphanumeric input mode Numeric input mode Multi-lingual uppercase and...
  • Page 31 Getting Started T46G T42G/T41P Description Conference The default contact icon The default caller photo This section describes how to configure basic network parameters for the IP phone. Note This section mainly introduces IPv4 network parameters. For more information on IPv6, refer to IPv6 Support on page 179.

  • Page 32: Dhcp

    Administrator’s Guide for SIP-T4X IP Phones Parameter DHCP Option Description Specify a list of IP addresses for routers on the Router client’s subnet. Specify a list of time servers available to the Time Server client. Domain Name Specify a list of domain name servers Server available to the client.
  • Page 33 Getting Started Phone User Interface Configure DHCP on the IP phone. To configure DHCP via web user interface: Click on Network->Basic. In the IPv4 Config block, mark the DHCP radio box. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after reboot. Click OK to reboot the IP phone.
  • Page 34 Administrator’s Guide for SIP-T4X IP Phones Procedure Network parameters can be configured manually using the configuration files or locally. Configure network parameters of the IP phone manually. Configuration File <MAC>.cfg For more information, refer to Static Network Settings on page 235.
  • Page 35 Getting Started To configure a static IPv4 address via web user interface: Click on Network->Basic. In the IPv4 Config block, mark the Static IP Address radio box. Enter the IP address, subnet mask, default gateway, primary DNS and secondary DNS in the corresponding fields. Click Confirm to accept the change.

  • Page 36: Pppoe

    Administrator’s Guide for SIP-T4X IP Phones PPPoE (Point-to-Point Protocol over Ethernet) is a network protocol used by Internet Service Providers (ISPs) to provide Digital Subscriber Line (DSL) high speed Internet services. PPPoE allows an office or building-full of users to share a common DSL connection to the Internet.
  • Page 37 The IP phone reboots automatically to make settings effective after a period of time. There are two Ethernet ports on the back of IP phones: Internet port and PC port. Three optional methods of transmission configuration for SIP-T4X IP phone Internet or PC Ethernet ports: Auto-negotiation ...
  • Page 38 Administrator’s Guide for SIP-T4X IP Phones Half-duplex Half-duplex transmission refers to transmitting voice or data in both directions, but in one direction at a time; this means one device can send data on the line, but not receive data simultaneously. You can configure the half-duplex transmission on both Internet port and PC port for IP phones to transmit in 10Mbps, 100Mbps or 1000Mbps (not applicable to SIP-T41P).

  • Page 39: Dial-Now

    Getting Started 239. Configure the transmission method of Ethernet port. Local Web User Interface Navigate to: http://<phoneIPAddress>/servlet ?p=network-adv&q=load To configure the transmission method of Ethernet port via web user interface: Click on Network->Advanced. Select the desired value from the pull-down list of WAN Port Link. Select the desired value from the pull-down list of PC Port Link.
  • Page 40 Administrator’s Guide for SIP-T4X IP Phones Block Out  You need to know the following basic regular expression syntax when creating dial plan: The dot “.” can be used as a placeholder or multiple placeholders for any string. Example: “12.” would match “123”, “1234”, “12345”, “12abc”, etc.
  • Page 41 Getting Started Procedure Replace rule can be created using the configuration files or locally. Create the replace rule for the IP phone. Configuration File <y0000000000xx>.cfg For more information, refer to Dial Plan on page 240. Create the replace rule for the IP phone.
  • Page 42 Administrator’s Guide for SIP-T4X IP Phones Dial-now is a string used to match the numbers entered by the user. When entered numbers match the predefined dial-now rule, IP phones will automatically dial out the numbers without employing the send key. IP phones support up to 100 dial-now rules, which can be created either one by one or in batch using a dial-now rule template.
  • Page 43 Getting Started If you leave the field blank or enter 0, the dial-now rule applies to all accounts on the IP phone. Click Add to add the dial-now rule. To configure the delay time for the dial-now rule via web user interface: Click on Features->General Information.
  • Page 44 Administrator’s Guide for SIP-T4X IP Phones 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 in front of the numbers and dial out.
  • Page 45 Getting Started If you leave the field blank or enter 0, the area code rule applies to all accounts on the IP phone. Click Confirm to accept the change. Block out rule prevents users from dialing out specific numbers. When entered numbers match the predefined block out rule, the LCD screen prompts “Forbidden Number”.
  • Page 46 Administrator’s Guide for SIP-T4X IP Phones the IP phone. Click Confirm to add the block out rule.
  • Page 47 Configuring Basic Features This chapter provides information for making configuration changes for the following basic features: Wallpaper  Backlight  User Password  Administrator Password  Phone Lock  Time and Date  Language  Logo Customization  Softkey Layout ...
  • Page 48 Administrator’s Guide for SIP-T4X IP Phones 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 49 Configuring Basic Features on page 356. Upload the customized wallpaper. Change the wallpaper via web Web User Interface user interface. Local Navigate to: http://<phoneIPAddress>/servlet ?p=settings-preference&q=load Change the wallpaper via phone Phone User Interface user interface. To upload customized wallpaper via web user interface: Click on Settings->Preference.
  • Page 50 Administrator’s Guide for SIP-T4X IP Phones Select the desired wallpaper from the pull-down list of Wallpaper. Click Confirm to accept the change. To change the wallpaper via phone user interface: Press Menu->Basic->Display->Wallpaper. Press , or the Switch soft key to select the desired wallpaper.
  • Page 51 Configuring Basic Features For more information, refer to Backlight on page 244. Configure the backlight of the LCD screen. Web User Interface Navigate to: http://<phoneIPAddress>/servlet Local ?p=settings-preference&q=load Configure the backlight of the Phone User Interface LCD screen. To configure the backlight via web user interface: Click on Settings->Preference.
  • Page 52 Administrator’s Guide for SIP-T4X IP Phones Some menu options are protected with two privilege levels, user and administrator, each with its own password. When logging into the web user interface, you need to enter the user name and password to access various menu options.
  • Page 53 Configuring Basic Features Advanced menu options are strictly for use by administrators. Users can configure them only if they have administrator privileges. The administrator password can only be changed by an administrator. The default administrator password is “admin”. For security reasons, the administrator should change the default administrator password as soon as possible.
  • Page 54 Administrator’s Guide for SIP-T4X IP Phones Click Confirm to accept the change. To change the administrator password via phone user interface: Press Menu->Advanced (password: admin) ->Set Password. Enter the current administrator password in the Current Password field. Enter a new administrator password in the New Password field and Confirm Password field.
  • Page 55 Configuring Basic Features http://<phoneIPAddress>/servl et?p=features-phonelock&q=lo Assign a keypad lock key. Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&model=1&q=loa d&linepage=1 Configure the phone lock type. 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 Enable.
  • Page 56 Administrator’s Guide for SIP-T4X IP Phones In the desired DSS key field, select Keypad Lock from the pull-down list of Type. Click Confirm to accept the change. To configure phone lock type via phone user interface: Press Menu->Advanced (password: admin) ->Phone Settings->Keypad Lock.

  • Page 57: Time Zone

    Configuring Basic Features Time Zone A time zone is a region on Earth that has a uniform standard time. It is convenient for areas in close commercial or other communication to keep the same time. When configuring IP phones to obtain the time and date from the NTP server, you must set the time zone.
  • Page 58 Administrator’s Guide for SIP-T4X IP Phones formats. For more information, refer to Time and Date on page 248. Configure the NTP server, time zone and DST. Configure the time and date manually. Web User Interface Configure the time and date formats.
  • Page 59 Configuring Basic Features Enter the end time in the End Date field. 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 60 Administrator’s Guide for SIP-T4X IP Phones Select Enabled from the pull-down list of Manual Time. Enter the time and date in the corresponding fields. Click Confirm to accept the change. To configure the time and date format via web user interface: Click on Settings->Time &...
  • Page 61 Configuring Basic Features The default time zone is "+8 China(Beijing)". Enter the domain names or IP addresses in the NTP Server 1 and NTP Server 2 fields, respectively. Press or the Switch soft key to select Automatic from the Daylight Saving field.
  • Page 62 Administrator’s Guide for SIP-T4X IP Phones Not all of the supported languages are available for selection. Languages available for selection depend on language packs currently loaded on IP phones. You can make languages available for use on the phone user interface by loading language packs to the IP phone.
  • Page 63 Configuring Basic Features Procedure Specify the language for the web user interface or the phone user interface using the configuration files or locally. Specify the languages for the phone user interface and the web user interface. Configuration File <y0000000000xx>.cfg For more information, refer to Language on page 253.
  • Page 64 Administrator’s Guide for SIP-T4X IP Phones Logo customization allows unifying the IP phone appearance or displaying a custom image on the idle screen such as a company logo, instead of the default system logo. Logo is not applicable to the SIP-T46G IP phone. The logo file format must be .dob, and the resolution of the SIP-T42G/T41P IP phones is 192*64 graphic.
  • Page 65 Configuring Basic Features Select Custom logo from the pull-down list of Use Logo. Click Browse to select the logo file from your local system. Click Upload to upload the file. Click Confirm to accept the change. The custom logo screen and the idle screen alternately display. Softkey layout is used to customize the soft keys at the bottom of the LCD screen to best suit the user needs.
  • Page 66 Administrator’s Guide for SIP-T4X IP Phones Call State Default Soft Key Optional Soft Key Reject Empty Empty Empty Switch Connecting Empty Cancel Connecting Transfer Empty Empty Switch SemiAttendTrans Empty Cancel Send Empty History Delete Directory Cancel Switch Dialing Line Directory...
  • Page 67 Configuring Basic Features Call State Default Soft Key Optional Soft Key Resume Switch NewCall Answer Cancel Reject Empty Empty Empty Switch Held Empty Answer Cancel Reject NewCall Transfer Empty Directory PreTrans Delete Switch Cancel Send Empty Empty Empty Switch InConference Empty Cancel Empty...
  • Page 68 Administrator’s Guide for SIP-T4X 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 69 Configuring Basic Features as Send on page 256. Configure the send key. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Web User Interface Configure send tone. Local Navigate to: http://<phoneIPAddress>/servlet ?p=features-audio&q=load Phone User Interface Configure the send key. To configure the send key via web user interface: Click on Features->General Information.
  • Page 70 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Send Sound. Click Confirm to accept the change. To configure the send key via phone user interface: Press Menu->Call Features->Others->General. Press , or the Switch soft key to select # or * from the Key as Send field, or select Disabled to disable this feature.
  • Page 71 Configuring Basic Features Configure the hotline number. Specify the time (in seconds) the IP phone waits to automatically Web User Interface dial out the hotline 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 to automatically dial out the hotline number.
  • Page 72 Administrator’s Guide for SIP-T4X IP Phones Call log contains call information such as remote party identification, time and date, and call duration. IP phones maintain a local call log. Call log consists of four lists: Missed calls, Placed calls, Received calls and Forwarded calls. Each call log list supports up to 100 entries.
  • Page 73 Configuring Basic Features To configure the call log via phone user interface: Press Menu->Call Features->Others->General. Press , or the Switch soft key to select the desired value from the History Record field. Press the Save soft key to accept the change. Missed call log allows IP phones to display the number of the missed calls with an indicator icon on the idle screen, and to log the missed calls in the missed calls list when IP phones miss calls.
  • Page 74 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Missed Call Log. Click Confirm to accept the change. The IP phone maintains a local directory. The local directory can store up to 1000 contacts and 50 groups (including the default groups: All Contacts, Company, Family, Friend and Blacklist).
  • Page 75 Configuring Basic Features To add a new group to the local directory via web user interface: Click on Directory->Local Directory. In the Group Setting block, enter the new group name in the Group field. Select the desired group ring tone from the pull-down list of Ring. Click Add to add the new group.
  • Page 76 Administrator’s Guide for SIP-T4X IP Phones It is not applicable to SIP-T42G and SIP-T41P IP phones. Click Add to add the contact. To add a group to the local directory via phone user interface: Press Menu->Directory->Local Group. Press the Add Group soft key.
  • Page 77 Configuring Basic Features Live dialpad allows IP phones to automatically dial out the entered phone number after a specified period of time. Procedure Live dialpad can be configured using the configuration files or locally. Configure live dialpad. Configuration File <y0000000000xx>.cfg For more information, refer to Live Dialpad...
  • Page 78 Administrator’s Guide for SIP-T4X IP Phones during conversation. Call waiting tone works only if call waiting is enabled. The call waiting on code and call waiting off code configured on IP phones are used to activate/deactivate the server-side call waiting feature. They may vary on different servers.
  • Page 79 Configuring Basic Features Select the desired value from the pull-down list of Call Waiting Tone. Click Confirm to accept the change. To configure call waiting and call waiting tone via phone user interface: Press Menu->Call Features->Call Waiting. Press , or the Switch soft key to select the desired value from the Call Waiting field.
  • Page 80 Administrator’s Guide for SIP-T4X IP Phones ?p=features-general&q=load Phone User Interface Configure auto redial. To configure auto redial via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Auto Redial. Enter the desired time interval (in seconds) in the Auto Redial Interval (1~300s) field.
  • Page 81 Configuring Basic Features for the IP phone to automatically answer a call. Procedure Auto answer can be configured using the configuration files or locally. Configure auto answer. <MAC>.cfg For more information, refer to Auto Answer on page 262. Configuration File Specify a period of delay time for auto answer.
  • Page 82 Administrator’s Guide for SIP-T4X IP Phones To configure a period of delay time for auto answer via web user interface: Click on Features->General Information. Enter the desired time (in seconds) in the Auto-Answer Delay(1~4s) field. Click Confirm to accept the change.
  • Page 83 Configuring Basic Features Procedure Call completion can be configured using the configuration files or locally. Configure call completion. Configuration File <y0000000000xx>.cfg For more information, refer to Call Completion on page 263. Configure call completion. Navigate to: Web User Interface http://<phoneIPAddress>/servlet Local ?p=features-general&q=load Phone User Interface...
  • Page 84 Administrator’s Guide for SIP-T4X IP Phones Example of anonymous SIP header: Via: SIP/2.0/UDP 10.2.8.183:5063;branch=z9hG4bK1535948896 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=128043702 To: <sip:1011@10.2.1.199> Call-ID: 1773251036@10.2.8.183 CSeq: 1 INVITE Contact: <sip:1012@10.2.8.183:5063> Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER,...
  • Page 85 Configuring Basic Features (Optional.) Enter the anonymous call on code in the On Code field. (Optional.) Enter the anonymous call off code in the Off Code field. Click Confirm to accept the change. To configure the anonymous call via phone user interface: Press Menu->Call Features->Anonymous.
  • Page 86 Administrator’s Guide for SIP-T4X IP Phones For more information, refer to Anonymous Call Rejection page 265. Configure anonymous call rejection. Navigate to: Web User Interface http://<phoneIPAddress>/servlet Local ?p=account-basic&q=load&acc Configure anonymous call Phone User Interface rejection. To configure anonymous call rejection via web user interface: Click on Account.
  • Page 87 Configuring Basic Features Press the Save soft key to accept the change. Do Not Disturb (DND) allows IP phones to ignore incoming calls. DND can be configured on a phone or per-line basis depending on the DND mode. Two DND modes: Phone (default): DND feature is effective for all accounts on the IP phone.
  • Page 88 Administrator’s Guide for SIP-T4X IP Phones page=1 Configure DND. Navigate to: http://<phoneIPAddress>/servlet? p=features-forward&q=load Specify return code and reason of the SIP response message. Navigate to: http://<phoneIPAddress>/servlet? p=features-general&q=load Assign a DND key. Phone User Interface Configure DND. To configure a DND key via web user interface: Click on DSSKey->Line Key.
  • Page 89 Configuring Basic Features 3) (Optional.) Enter the DND off code in the DND Off Code field. b) If you mark the Custom radio box: 1) Select the desired account from the pull-down list of Account. 2) Mark the desired value in the DND Status field. 3) (Optional.) Enter the DND on code in the DND On Code field.
  • Page 90 Administrator’s Guide for SIP-T4X IP Phones To specify the return code via web user interface: Click on Features->General Information. Select the desired type from the pull-down list of Return Code When DND. Click Confirm to accept the change. To configure a DND key via phone user interface: Press Menu->Call Features->DSS Keys.
  • Page 91 Configuring Basic Features Press the All On soft key to activate DND for all accounts. Press the Save soft key to accept the change. Busy tone is audible to the other party, indicating that the call connection has been broken when one party releases a call. Busy tone delay can define a period of time during which the busy tone is audible.
  • Page 92 Administrator’s Guide for SIP-T4X IP Phones Return code when refuse defines the return code and reason of the SIP response message for call rejection. The caller’s LCD screen displays the reason according to the return code received. Available return codes and reasons are: 404 (Not found) ...
  • Page 93 Configuring Basic Features Early media refers to media (e.g., audio and video) played to the caller before a SIP call is actually established. Current implementation supports early media through the 183 message. When the caller receives a 183 message with SDP before the call is established, a media channel is established.
  • Page 94 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of 180 Ring Workaround. Click Confirm to accept the change. An outbound proxy server can receive all initiating request messages and route them to the designated destination. If the IP phone is configured to use an outbound proxy server within a dialog, all SIP request messages from the IP phone will be sent to the outbound proxy server forcefully.
  • Page 95 Configuring Basic Features ?p=features-general&q=load To specify whether to use outbound proxy server in a dialog via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Use Outbound Proxy in Dialog. Click Confirm to accept the change. SIP session timers T1, T2 and T4 are SIP transaction layer timers defined in RFC 3261.
  • Page 96 Administrator’s Guide for SIP-T4X IP Phones ?p=account-adv&q=load&acc= 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.
  • Page 97 Configuring Basic Features 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 273. Configure session timer. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=account-adv&q=load&acc= To configure session timer via web user interface:...
  • Page 98 Administrator’s Guide for SIP-T4X IP Phones Call hold provides a service of putting 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 99 Configuring Basic Features Select the desired value from the pull-down list of RFC 2543 Hold. Click Confirm to accept the change. To configure call hold tone and call hold tone delay via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Play Hold Tone. Enter the desired time in the Play Hold Tone Delay field.
  • Page 100 Administrator’s Guide for SIP-T4X IP Phones 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 calls immediately.
  • Page 101 Configuring Basic Features vlet?p=features-forward&q=l Configure forward 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. In the Forward block, mark the desired radio box in the Mode field. a) If you mark the Phone radio box: 1) Mark the desired radio box in the Always Forward/Busy Forward/No Answer Forward field.
  • Page 102 Administrator’s Guide for SIP-T4X IP Phones 3) (Optional.) 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 Times (only for no answer forward).
  • Page 103 Configuring Basic Features To configure call forward in phone mode via phone user interface: Press Menu->Call 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: 1) Press , or the Switch soft key to select the desired value from the Always Forward field.
  • Page 104 Administrator’s Guide for SIP-T4X IP Phones Always Forward field. 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.
  • Page 105 Configuring Basic Features 3) Press the OK soft key to accept the change. Press the Save soft key to accept the change. 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 106 Administrator’s Guide for SIP-T4X IP Phones To configure call transfer via web user interface: 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.
  • Page 107 Configuring Basic Features To configure the network conference via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced. Select Network from the pull-down list of Conference Type. Enter the conference URI in the Conference URI field. Click Confirm to accept the change.
  • Page 108 Administrator’s Guide for SIP-T4X IP Phones hang up. Navigate to: http://<phoneIPAddress>/servlet ?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.
  • Page 109 Configuring Basic Features Directed Call Pickup on page 288. Assign a directed call pickup key. For more information, refer to Directed Call Pickup Key page 362. <y0000000000xx>.cfg Configure the directed call pickup feature on a phone basis. For more information, refer to Directed Call Pickup on page 287.
  • Page 110 Administrator’s Guide for SIP-T4X IP Phones Select the desired line from the pull-down list of Line. Click Confirm to accept the change. To configure the directed call pickup feature on a phone basis via web user interface: Click on Features->Call Pickup.
  • Page 111 Configuring Basic Features 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->Call Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field.
  • Page 112 Administrator’s Guide for SIP-T4X IP Phones 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 289.
  • Page 113 Configuring Basic Features Select the desired line from the pull-down list of Line. Click Confirm to accept the change. To configure the group call pickup feature on a phone basis via web user interface: Click on Features->Call Pickup. 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 114 Administrator’s Guide for SIP-T4X 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->Call Features->DSS Keys. Select the desired DSS key.
  • Page 115 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 116 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Dialog Info Call Pickup. Click Confirm to accept the change. Call return, also known as last call return, allows users to place a call back to the last caller.
  • Page 117 Configuring Basic Features In the desired DSS 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->Call Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field.
  • Page 118 Administrator’s Guide for SIP-T4X IP Phones d&linepage=1 Phone User Interface Assign a call park key. To configure a call park key via web user interface: Click on DSSKey->Line Key. In the desired DSS key field, select Call Park from the pull-down list of Type.

  • Page 119: Web Server Type

    Configuring Basic Features Procedure Web server type can be configured using the configuration files or locally. Specify the web access type, HTTP port and HTTPS port. Configuration File <y0000000000xx>.cfg For more information, refer to Web Server Type on page 290. Specify the web access type, HTTP port and HTTPS port.

  • Page 120: Calling Line Identification Presentation

    Administrator’s Guide for SIP-T4X IP Phones A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the IP phone. To configure the web server type via phone user interface: Press Menu->Advanced (password: admin) ->Network->Webserver Type.

  • Page 121: Connected Line Identification Presentation

    Configuring Basic Features To configure the presentation of the caller identity 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 the Caller ID Source. Click Confirm to accept the change.

  • Page 122: Dtmf

    Administrator’s Guide for SIP-T4X IP Phones DTMF (Dual Tone Multi-frequency), better known as touch-tone, is used for telecommunication signaling over analog telephone lines in the voice-frequency band. DTMF is the signal sent from the IP phone to the network, which is generated when pressing the IP phone’s keypad during a call.
  • Page 123 Configuring Basic Features same VoIP codec as your voice and is audible to the 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 124 Administrator’s Guide for SIP-T4X IP Phones If SIP INFO or AUTO or SIP INFO is selected, select the desired value from the 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.

  • Page 125: Suppress Dtmf Display

    Configuring Basic Features Suppress DTMF display allows IP phones to suppress the display of DTMF digits. The 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 126: Transfer Via Dtmf

    Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Suppress DTMF Display Delay. Click Confirm to accept the change. Call transfer is implemented via DTMF on some traditional servers. The IP phone sends specified DTMF digits to the server for transferring calls to a third party.
  • Page 127 Configuring Basic Features Enter the specified DTMF digits in the Tran Send DTMF field. Click Confirm to accept the change. Intercom allows establishing an audio conversation directly. The IP phone can answer intercom calls automatically. This feature depends on support from a SIP server. Intercom is a useful feature in office environments to quickly connect with an operator or secretary.

  • Page 128: Intercom

    Administrator’s Guide for SIP-T4X IP Phones nepage=1 Phone User Interface Assign an intercom key. To configure an intercom key via web user interface: Click on DSSKey->Line Key. In the desired DSS key field, select Intercom from the pull-down list of Type.

  • Page 129: Incoming Intercom Calls

    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 130 Administrator’s Guide for SIP-T4X 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->Call Features->Intercom.

  • Page 131: 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  Message Waiting Indicator ...
  • Page 132 Administrator’s Guide for SIP-T4X IP Phones ring tone. Alert-Info headers in the following two formats: Alert-Info: 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  the Bellcore-drN ring tone (N=1,2,3,4,5). Example: Alert-Info: http://127.0.0.1/Bellcore-dr1...
  • Page 133 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 plays the local ring tone associated with info text.
  • Page 134 Administrator’s Guide for SIP-T4X 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 135 Configuring Advanced Features When receiving a message or recording a call, 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.
  • Page 136 Administrator’s Guide for SIP-T4X IP Phones Configured tones can be heard on the IP phone for the following conditions: Condition Description Dial When in the pre-dialing interface Ring Back Ring-back tone Busy When the callee is busy Congestion When the network is congested...
  • Page 137 Configuring Advanced Features If you select Custom, you can customize the tone for indicating each condition of the IP phone. Click Confirm to accept the change. Remote phone book is the centrally maintained phone book, stored on the remote server. Users only need the access URL of the remote phone book. The IP phone can establish a connection with the remote server and download the entries, and then display the phone book entries on the phone user interface.
  • Page 138 Administrator’s Guide for SIP-T4X IP Phones receives incoming calls. Specify how often the IP phone refreshes the local cache of the remote phone book. For more information, refer to Remote Phone Book on page 302. Specify the access URL of the remote phone book.
  • Page 139 Configuring Advanced Features To configure the remote phone book via web user interface: Click on Directory->Remote Phone Book. Select the desired value from the pull-down list of Search Remote Phonebook Name. Enter the desired time in the Search Flash Time (Seconds) field. Click Confirm to accept the change.
  • Page 140 Administrator’s Guide for SIP-T4X 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 commonName from given name joined to surname.
  • Page 141 Configuring Advanced Features To configure LDAP via web user interface: Click on Directory->LDAP. Select Enabled from the pull-down list of Enable LDAP. Enter the values in the corresponding fields. Select the desired values from the corresponding pull-down lists. Click Confirm to accept the change. To configure an LDAP key via web user interface: Click on DSSKey->Line Key.
  • Page 142 Administrator’s Guide for SIP-T4X IP Phones Press , or the Switch soft key to select Key Event from the Type field. Press , or the Switch soft key to select LDAP from the Key Event field. (Optional.) Enter the string that will appear on the LCD screen in the Label field.
  • Page 143 Configuring Advanced Features Line key LED (configured as BLF key when LED Off in Idle is enabled) LED Status Description The monitored user is busy. Solid red The call is parked against the monitored user’s phone number. The monitored user receives an incoming call. Fast flashing red The monitored user is idle.
  • Page 144 Administrator’s Guide for SIP-T4X IP Phones To configure a BLF key via web user interface: Click on DSSKey->Line Key. In the desired DSS key field, select BLF from the pull-down list of Type. Enter the phone number or extension you want to monitor in the Value field.
  • Page 145 Configuring Advanced Features To configure the LED off in idle via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of LED Off in Idle. Click Confirm to accept the change. To configure a BLF key via phone user interface: Press Menu->Call Features->DSS Keys.
  • Page 146 Administrator’s Guide for SIP-T4X IP Phones Internet) to the held party. Procedure Music on Hold can be configured using the configuration files or locally. Configure the MoH feature on a per-line basis. Configuration File <MAC>.cfg For more information, refer to Music on Hold on page 310.
  • Page 147 Configuring Advanced Features 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. ACD depends on support from a SIP server.
  • Page 148 Administrator’s Guide for SIP-T4X IP Phones Phone User Interface Assign an ACD key. To configure an ACD key via web user interface: Click on DSSKey->Line Key. In the desired DSS key field, select ACD from the pull-down list of Type.
  • Page 149 Configuring Advanced Features To configure an ACD key via phone user interface: Press Menu->Call Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select ACD from the Type field. (Optional.) Enter the string that will appear on the LCD screen in the Label field. Press the Save soft key to accept the change.
  • Page 150 Administrator’s Guide for SIP-T4X IP Phones To configure subscribe for MWI via web user interface: Click on Account. Select the desired account from the pull-down list of Account. Click on Advanced. Select the desired value from the pull-down list of Subscribe for MWI.
  • Page 151 Configuring Advanced Features Enter the desired voice number in the Voice Mail field. Click Confirm to accept the change. Multicast paging allows IP phones to send/receive Real-time Transport Protocol (RTP) streams to/from the pre-configured multicast address(es) without involving SIP signaling. Up to 10 listening multicast addresses can be specified on the IP phone.

  • Page 152: Sending Rtp Stream

    Administrator’s Guide for SIP-T4X IP Phones RTP . For more information, refer to Sending RTP Stream on page 313. Assign a multicast paging key. Navigate to: http://<phoneIPAddress>/servlet ?p=dsskey&model=1&q=load&li nepage=1 Web User Interface Specifies a multicast codec for Local the IP phone to use to send the RTP stream.
  • Page 153 Configuring Advanced Features 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->Call Features->DSS Keys. Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field.
  • Page 154 Administrator’s Guide for SIP-T4X IP Phones multicast paging calls with higher priority are automatically answered and the ones with lower priority are ignored. Paging Priority Active This parameter decides how the IP phone handles the incoming multicast paging calls when there is already a multicast paging call in progress. If the parameter is configured as disabled, the IP phone will automatically ignore all incoming multicast paging calls.
  • Page 155 Configuring Advanced Features The label will appear on the LCD screen when receiving the RTP multicast. Click Confirm to accept the change. To configure the paging barge and paging priority active features via web user interface: Click on Directory->Multicast IP. Select the desired value from the pull-down list of Paging Barge.

  • Page 156: Call Recording

    Administrator’s Guide for SIP-T4X IP Phones Call recording enables users to record calls. It depends on support from a SIP server. When the user presses the call record key, the IP phone sends a record request to the server. 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.
  • Page 157 Get /phonerecording.cgi?model=yealink HTTP/1.0\r\n Request Method: GET Request URI: /phonerecording.cgi?model=yealink Request version: HTTP/1.0 Host: 10.1.2.224\r\n User-agent: yealink SIP-T46G 28.71.0.10 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 158 Administrator’s Guide for SIP-T4X IP Phones <Text> The recording session is successfully stopped. </Text> <YealinkIPPhoneText> 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 370.

  • Page 159: Hot Desking

    Configuring Advanced Features To configure a URL record key via web user interface: Click on DSSKey->Line Key. In the desired DSS key field, select URL Record from the pull-down list of Type. Enter the URL in the Value field. Click Confirm to accept the change. To configure a record key via phone user interface: Press Menu->Call Features->DSS Keys.
  • Page 160 Administrator’s Guide for SIP-T4X IP Phones time, which means actual personal offices would often be vacant, consuming valuable space and resources. The hot desking feature allows a user to clear registration configurations of all accounts on the IP phone, and then register his account on line 1. To use this feature, you need to assign a hot desking key.
  • Page 161 Configuring Advanced Features field. (Optional.) Enter the string that will appear on the LCD screen in the Label field. Press the Save soft key to accept the change. Action URL allows IP phones to interact with web server applications by sending an HTTP or HTTPS GET request.
  • Page 162 Administrator’s Guide for SIP-T4X IP Phones Event Description UnHold When the IP phone retrieves a hold call. Mute When the IP phone mutes a call. UnMute When the IP phone unmutes a call. Missed Call When the IP phone misses a call.

  • Page 163: Action Url

    Configuring Advanced Features Variable Value Description call or establishes a call. The SIP URI of the caller when the IP phone places a call. $local 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.
  • Page 164 Administrator’s Guide for SIP-T4X IP Phones Enter the action URLs in the corresponding fields. Click Confirm to accept the change. Opposite to action URL, action URI allows IP phones to interact with web server application by receiving and handling an HTTP or HTTPS GET request. When receiving a GET request, the IP phone will perform the specified action and respond with a 200 OK message.
  • Page 165 Configuring Advanced Features Variable Value Phone Action CANCEL Return to a previous screen or cancel a call. 0-9/*/POUND Send the DTMF digit (0-9, * or #). Press the line key (for SIP-T46G, X=27, for L1-LX SIP-T42G/T41P , X=15). F_CONFERENCE Press the Conference soft key. F1-F4 Press the soft key.

  • Page 166: Action Uri

    Administrator’s Guide for SIP-T4X IP Phones addresses on the IP phone, or configure the IP phone to receive and handle the URI from any IP address. Procedure Specify the trusted IP address for Action URI using the configuration files or locally.

  • Page 167: Server Redundancy

    Configuring Advanced Features 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. Two types of redundancy are possible.

  • Page 168: Phone Registration

    Administrator’s Guide for SIP-T4X IP Phones The primary server is the highest priority server in a cluster of servers resolved by the DNS server. The secondary server backs up a primary server when the primary server fails and offers the same functionality as the primary server.
  • Page 169 Configuring Advanced Features and service type (UDP , TCP and TLS), the SRV query on the record returned from the NAPTR for the host name and the port number, and the A query for the IP addresses. If a port is set to 0 and the transport type is set to DNS-NAPTR, NAPTR and SRV queries will be tried before falling back to A query.
  • Page 170 Administrator’s Guide for SIP-T4X IP Phones Parameters are explained in the following table: Parameter Description Specify preferential treatment for the specific record. The order order is from lowest to highest, lower order is MORE preferred. Specify the preference for processing multiple NAPTR records pref with the same order value.
  • Page 171 Configuring Advanced Features SRV query returns two records. The two SRV records point to different hosts and have the same priority 0. The weight of the second record is higher than the first one, so the second record will be picked first. The two records also contain a port “5060”, the IP phone uses this port.
  • Page 172 Administrator’s Guide for SIP-T4X IP Phones Configure the transport type. Navigate to: Local Web User Interface http://<phoneIPAddress>/servl et?p=account-register&q=load &acc=0 To configure the server redundancy and transport type via web user interface: Click on Account. Select the desired account from the pull-down list of Account.

  • Page 173: Lldp

    The default value is 60s. End of LLDPDU Marks end of LLDPDU. Name assigned to the IP phone. System Name The default value is “yealink”. Description of the IP phone. System Description The default value is “yealink”. The supported and enabled phone capabilities.
  • Page 174 Administrator’s Guide for SIP-T4X IP Phones TLV Type TLV Name Description The advertised capabilities of PMD. Auto-Negotiation is: 100BASE-TX (full duplex mode), 100BASE-TX (half duplex mode), 10BASE-T (full duplex mode), 10BASE-T (half duplex mode). The MED device type of the IP phone and the supported LLDP-MED TLV type can be encapsulated in LLDPDU.

  • Page 175: Vlan

    Configuring Advanced Features Navigate to: http://<phoneIPAddress>/servl et?p=network-adv&q=load To configure LLDP via web user interface: Click on Network->Advanced. In the LLDP block, select the desired value from the pull-down list of Active. Enter the desired time interval in the Packet Interval (1~3600s) field. Click Confirm to accept the change.
  • Page 176 Administrator’s Guide for SIP-T4X IP Phones The VLAN feature on IP phones allows simultaneous access for a regular PC. This feature allows a PC to be daisy chained to an IP phone and the connection for both PC and IP phone to be trunked through the same physical Ethernet cable.
  • Page 177 Configuring Advanced Features Select the desired value (0-7) from the pull-down list of Priority. Click Confirm to accept the change. A dialog box pops up to prompt reboot to make the settings effective. Click OK to reboot the IP phone. To configure VLAN for PC port via web user interface: Click on Network->Advanced.

  • Page 178: Vpn

    Administrator’s Guide for SIP-T4X IP Phones To configure the 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.
  • Page 179 .tar. The VPN-related files are: certificates (ca.crt and client.crt), key (client.key) and the configuration file (vpn.cnf) of the VPN client. For more information on how to VPN Feature on Yealink IP Phones package a .tar file, refer to Procedure VPN can be configured using the configuration files or locally.
  • Page 180 Administrator’s Guide for SIP-T4X IP Phones Click Browse to locate the tar package 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.

  • Page 181: Voice Qos

    Configuring Advanced Features QoS provides better network service by providing the following features: 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 the IP networks. It provides no guarantees for data delivering, which means delay, jitter, packet loss and bandwidth allocation are unpredictable.
  • Page 182 Administrator’s Guide for SIP-T4X IP Phones SIP QoS 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.
  • Page 183 Configuring Advanced Features Enter the desired value in the SIP Qos (0~63) field. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the IP phone. Network Address Translation (NAT) is essentially a translation table that maps public IP address and port combinations to private ones.
  • Page 184 Administrator’s Guide for SIP-T4X IP Phones assistance from a third-party network server (STUN server) usually located on public Internet. The IP phone can be configured to act as a STUN client, sending exploratory STUN messages to the STUN server. The STUN server uses those messages to determine the public IP address and port used, and then informs the client.
  • Page 185 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, together with the contact information. YEALINK-MIB 1.3.6.1.2.1.37459.2.1.1.0 For example, Sysadmin (root@localhost) An administratively-assigned name for YEALINK-MIB 1.3.6.1.2.1.37459.2.1.2.0...

  • Page 186: Snmp

    Administrator’s Guide for SIP-T4X IP Phones Description MacVersion[0.0.0.1]ComVersion[0.0.0.1] Procedure SNMP can be configured using the configuration files or locally. Configure SNMP on the IP phone. Configuration File <y0000000000xx>.cfg For more information, refer to SNMP on page 329. Configure SNMP .
  • Page 187 Configuring Advanced Features A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the IP phone. IEEE 802.1X authentication is an IEEE standard for Port-based Network Access Control (PNAC), part of the IEEE 802.1 group of networking protocols. It offers an authentication mechanism for devices to connect to a LAN or WLAN.
  • Page 188 Administrator’s Guide for SIP-T4X IP Phones 2) Enter the password for authentication in the MD5 Password field. b) If you select EAP-TLS: 1) Enter the user name for authentication in the Identity field. 2) Leave the MD5 Password field blank.
  • Page 189 Configuring Advanced Features 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. 3) In the CA Certificates field, click Browse to locate the desired certificate (*.pem,*.crt, *.cer or *.der) from your local system.
  • Page 190 Administrator’s Guide for SIP-T4X IP Phones 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to locate the desired certificate (*.pem,*.crt, *.cer or *.der) from your local system.
  • Page 191 Configuring Advanced Features 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the Password field. Click Save to accept the change. The IP phone reboots automatically to make the settings effective after a period of time.
  • Page 192 Administrator’s Guide for SIP-T4X IP Phones RPC Method Description This method is used to cause the CPE to download a specified file from the designated location. File types supported by IP phones are: Download Firmware Image  Configuration File ...
  • Page 193 Configuring Advanced Features Enter the user name and password authenticated by the ACS in the ACS Username and ACS Password fields. 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.
  • Page 194 Administrator’s Guide for SIP-T4X IP Phones network with at least one IPv6 router connected. This router is configured by the network administrator and sends out Router Advertisement announcements onto the link. These announcements can allow the on-link connected IP phone to configure itself with IPv6 address, as specified in RFC 4862.
  • Page 195 Configuring Advanced Features If you mark the Static IP Address radio box, configure the IPv6 address and other configuration parameters in the corresponding fields. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the IP phone.
  • Page 196 Administrator’s Guide for SIP-T4X IP Phones In the ICMPv6 Status block, select the desired value from the pull-down list of Active. Click Confirm to accept the change. To configure IPv6 address via phone user interface: Press Menu->Advanced (password: admin) ->Network->WAN Port.

  • Page 197: Configuring Audio Features

    Configuring Audio Features This chapter provides information for making configuration changes for the following audio features: Headset Prior  Dual Headset  Audio Codecs  Acoustic Clarity Technology  Headset prior allows users to use headset preferentially if a headset is physically connected to the IP phone.
  • Page 198 Administrator’s Guide for SIP-T4X IP Phones Select the desired value from the pull-down list of Headset Prior. Click Confirm to accept the change. 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 199 Configuring Audio Features Select the desired value from the pull-down list of Dual-Headset. Click Confirm to accept the change. CODEC is an abbreviation of COmpress-DECompress, capable of coding or decoding a digital data stream or signal by implementing an algorithm. The object of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining the quality.
  • Page 200 Administrator’s Guide for SIP-T4X 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 201 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 339.
  • Page 202 Administrator’s Guide for SIP-T4X IP Phones Click to adjust the priority of the enabled codecs. Click Confirm to accept the change. To configure the Ptime on a per-line basis via web user interface: Click on Account. Select the desired account from the pull-down list of Account.

  • Page 203: Acoustic Echo Cancellation

    Configuring Audio Features 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 204: Voice Activity Detection

    Administrator’s Guide for SIP-T4X IP Phones 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 used to deactivate some processes during non-speech section of an audio session.

  • Page 205: Comfort Noise Generation

    Configuring Audio Features 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 206: Jitter Buffer

    Administrator’s Guide for SIP-T4X IP Phones 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 207 Configuring Audio Features Enter the fixed delay time for fixed jitter buffer in the Normal field. Click Confirm to accept the change.
  • Page 208 Administrator’s Guide for SIP-T4X IP Phones...

  • Page 209: Configuring Security Features

    SIP-T4X IP phones support TLS version 1.0. A cipher suite is a named combination of authentication, encryption, and message authentication code (MAC) algorithms used to negotiate the security settings for a network connection using the TLS/SSL network protocol.
  • Page 210 Administrator’s Guide for SIP-T4X IP Phones AES256-SHA  EDH-RSA-DES-CBC3-SHA  EDH-DSS-DES-CBC3-SHA  DES-CBC3-SHA  DHE-RSA-AES128-SHA  DHE-DSS-AES128-SHA  AES128-SHA  IDEA-CBC-SHA  DHE-DSS-RC4-SHA  RC4-SHA  RC4-MD5  EXP1024-DHE-DSS-DES-CBC-SHA  EXP1024-DES-CBC-SHA  EDH-RSA-DES-CBC-SHA  EDH-DSS-DES-CBC-SHA  DES-CBC-SHA  EXP1024-DHE-DSS-RC4-SHA ...
  • Page 211 Configuring Security Features negotiation with “Server Hello Done” message. Step3: The IP phone sends session key information (encrypted with server’s public key) in the “Client Key Exchange” message. Step4: Server sends “Change Cipher Spec” message to activate the negotiated options for all future messages it will send.
  • Page 212 Administrator’s Guide for SIP-T4X IP Phones Configure the trusted certificates feature. Configure the server certificates feature. For more information, refer to on page 345. <y0000000000xx>.cfg Upload the trusted certificates. Upload the server certificates. For more information, refer to Uploading Certificates on page 345.
  • Page 213 Configuring Security Features Select the desired value from the pull-down list of CA Certificates. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the IP phone. To configure TLS on a per-line basis via web user interface: Click on Account.
  • Page 214 Administrator’s Guide for SIP-T4X IP Phones Click Browse to locate the certificate (*.pem,*.crt, *.cer or *.der) from your local system. Click Upload to upload the certificate. To configure the server certificates feature via web user interface: Click on Security->Server Certificates.
  • Page 215 Configuring Security Features Click Browse to locate the certificate (*.pem or *.cer) from your local system. Click Upload to upload the certificate. The dialog box pops up to prompt “Success: The Server Certificate has been loaded! Rebooting, please wait…”. Secure Real-Time Transport Protocol (SRTP) encrypts RTP streams during VoIP phone calls to avoid interception and eavesdropping.
  • Page 216 Administrator’s Guide for SIP-T4X IP Phones The callee receives the INVITE message with the RTP encryption algorithm, and then answers the call by responding with a 200 OK message which carries the negotiated RTP encryption algorithm. Example of the RTP encryption algorithm carried in the SDP of the 200 OK message:...
  • Page 217 This tool generates another new file named as Aeskey.txt to store the plaintext 16-character symmetric keys for each configuration file. For a Microsoft Windows platform , you can use Yealink-supplied encryption tool "Config_Encrypt_Tool.exe" to encrypt the <y0000000000xx>.cfg and <MAC>.cfg files respectively.

  • Page 218: Procedure To Encrypt Configuration Files

    Administrator’s Guide for SIP-T4X 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 219 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 and AES keys can be configured using the configuration files or locally.
  • Page 220 Administrator’s Guide for SIP-T4X IP Phones To configure the AES keys via web user interface: Click on Settings->Auto Provision. Enter the values in the Common AES Key and MAC-Oriented AES Key fields. Click Confirm to accept the change.

  • Page 221: Upgrading Firmware

    36.x.x.x.rom SIP-T41P 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 the local system in advance. To upgrade firmware manually via web user interface: Click on Settings->Upgrade.
  • Page 222 Administrator’s Guide for SIP-T4X 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 upgrade. Note Do not unplug the network and power cables when the IP phone is upgrading firmware.
  • Page 223 Upgrading Firmware For more information, refer to Upgrading Firmware on page 350. 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 224 Administrator’s Guide for SIP-T4X IP Phones...

  • Page 225: Resource Files

    IP phone. 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 226 Administrator’s Guide for SIP-T4X 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 227 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 228 Administrator’s Guide for SIP-T4X IP Phones the 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 229 Resource Files 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 230 Administrator’s Guide for SIP-T4X IP Phones Where: display_name=”” specifies the name of the contact (This value cannot be blank or duplicated). office_number =”” specifies the office number of the contact. mobile_number=”” specifies the mobile number of the contact. other_number=”” specifies the other number of the contact.
  • Page 231 <DirectoryEntry> <Name>Jack</Name> <Telephone>1003</Telephone> </DirectoryEntry> <DirectoryEntry> <Name>John</Name> <Telephone>1004</Telephone> </DirectoryEntry> <DirectoryEntry> <Name>Marry</Name> <Telephone>1005</Telephone> </DirectoryEntry> </YealinkIPPhoneDirectory> Note Yealink supplies a phone book generation tool to quickly generate a remote XML phone Yealink Phonebook Generation Tool User Guide book. For more information, refer to...

  • Page 232: Remote Xml Phonebook

    Administrator’s Guide for SIP-T4X IP Phones Access URL of the resource file can be configured in the configuration files: Configure the access URL of the replace rule template. Configuration File <y0000000000xx>.cfg For more information, refer to Access URL of Replace Rule Template on page 352.

  • Page 233: Troubleshooting

    This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using SIP-T4X IP phones. IP phones can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which can help an administrator more easily find the system problem and fix it.
  • Page 234 Administrator’s Guide for SIP-T4X 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?”. Click OK to reboot the IP phone.
  • Page 235 Troubleshooting Click Confirm to accept the change. A dialog box pops up to prompt “Do you want to restart your machine?”. Click OK to reboot the IP phone. The system log will be exported to the desired syslog server after reboot. To export a log file to the local system via web user interface: Click on Settings->Configuration.
  • Page 236 Administrator’s Guide for SIP-T4X IP Phones You can capture packets in two ways: capturing the packets via web user interface or using the Ethernet software. You can analyze the packets captured for troubleshooting purpose. To capture packet via web user interface: Click on Settings->Upgrade.
  • Page 237 Troubleshooting To configure watch dog via web user interface: Click on Settings->Preference. Select the desired value from the pull-down list of Watch Dog. Click Confirm to accept the change. Status indicators may consist of the power LED, line key indicator, headset key indicator mute key indicator and the on-screen icon or error messages.
  • Page 238 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.
  • Page 239 Troubleshooting Press the OK key when the IP phone is idle to check the basic information (e.g., IP address MAC address and firmware version). ’ Do one of the following: Ensure that the target firmware is not the same as the current firmware. ...
  • Page 240 Administrator’s Guide for SIP-T4X IP Phones A remote phone book is placed on a server, while a local phonebook is placed on the IP phone flash. A remote phone book can be used by everyone that can access the server, while a local phonebook can only be used by a specific phone.
  • Page 241 Troubleshooting IP phones use the PoE preferentially. Auto provisioning refers to the update of IP phones, including update on the configuration parameters, local phonebook, firmware and so on. You can use auto provisioning on a single phone, but it makes more sense in mass deployment. Plug and Play (PnP) is a method for IP phones to acquire the provisioning server address.
  • Page 242 Administrator’s Guide for SIP-T4X IP Phones Do one of the following: Reset another available IP address for the IP phone.  Check network configuration via phone user interface at the path  Menu->Advanced->Network->WAN Port->IPv4 (or IPv6). If the Static IP is selected, select DHCP instead.
  • Page 243 Appendix 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 offers an authentication mechanism for devices to connect to a LAN or WLAN. ACD (Automatic Call Distribution) — used to distribute calls from large volumes of incoming calls to the registered IP phone users.
  • Page 244 Administrator’s Guide for SIP-T4X IP Phones IEEE (Institute of Electrical and Electronics Engineers) — a non-profit professional association headquartered in New York City that is dedicated to advancing technological innovation and excellence. LAN (Local Area Network) — used to interconnects network devices in a limited area such as a home, school, computer laboratory, or office building.
  • Page 245 Appendix 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 Canada-Manitoba(Winnipeg) −06:00...
  • Page 246 Administrator’s Guide for SIP-T4X 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 247 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 248 Administrator’s Guide for SIP-T4X IP Phones This appendix describes configuration parameters in the configuration files for each feature. The configuration files are <y0000000000xx>.cfg and <MAC>.cfg. 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. The IP phone checks for configuration files and looks for resource files when restarting the IP phone.
  • Page 249 Appendix Parameter- Configuration File network.internet_port.type <MAC>.cfg Defines 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 Example network.internet_port.type = 2 Parameter-...
  • Page 250 Administrator’s Guide for SIP-T4X IP Phones port type is configured as Static IP Address. 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 network.internet_port.ip = 192.168.1.20...
  • Page 251 Appendix Parameter- Configuration File network.primary_dns <MAC>.cfg Configures the primary DNS server when the Internet port type is configured as Static IP Address. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value...
  • Page 252 Administrator’s Guide for SIP-T4X IP Phones Valid values are: 0-DHCP Range 1-PPPoE 2-Static IP Address Example network.internet_port.type= 1 Parameter- Configuration File network.pppoe.user <y0000000000xx>.cfg Configures the PPPoE user name when the Internet port type is configured as PPPoE. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 253 Appendix Internet Port Negotiation Parameter- Configuration File <y0000000000xx>.cfg network.internet_port.speed_d uplex Specifies the transmission method of Internet port. Description Note: We recommend that you do not change this parameter. Format Integer Default Value Valid values are: 0-Auto negotiate 1-Full duplex, 10Mbps 2-Full duplex, 100Mbps Range 3-Half duplex, 10Mbps...
  • Page 254 Administrator’s Guide for SIP-T4X IP Phones Example network.pc_port.speed_duplex = 0 Replace Rule Parameter- Configuration File dialplan.replace.prefix.x <y0000000000xx>.cfg Specifies the string you want to replace. Description X ranges from 1 to 100. Format String Default Value Blank Range Not Applicable Example dialplan.replace.prefix.1 = 91([5-7])12...
  • Page 255 Appendix 0 to 6 (for SIP-T46G) 0 to 3 (for SIP-T42G/T41P) Example dialplan.replace.line_id.1 = 1,2 Dial-now Parameter- Configuration File dialplan.dialnow.rule.x <y0000000000xx>.cfg Specifies the string used to match the numbers entered by the user. When entered numbers match the predefined dial-now rule, the IP Description phone will automatically dial out the numbers without pressing the send key.
  • Page 256 Administrator’s Guide for SIP-T4X IP Phones When entered numbers match the predefined dial-now rule, the IP phone will automatically dial out the entered number after the specified delay time. Format Integer Default Value Range 1 to 14 Example phone_setting.dialnow_delay = 1...
  • Page 257 Appendix Default Value Range 1 to 15 Example dialplan.area_code.max_len = 13 Parameter- Configuration File dialplan.area_code.line_id <y0000000000xx>.cfg Specifies the desired line to apply this area code rule. Description Note: Multiple line IDs are separated by comma. Format Integer Default Value Blank (for all lines) Valid values are: Range 0 to 6 (for SIP-T46G)
  • Page 258 Administrator’s Guide for SIP-T4X IP Phones Format Integer Default Value Blank (for all lines) Valid values are: Range 0 to 6 (for SIP-T46G) 0 to 3 (for SIP-T42G/T41P) Example dialplan.block_out.line_id.1 = 1,2,3 Parameter- Configuration File <y0000000000xx>.cfg phone_setting.active_backlight _level Configures the backlight level used to adjust the backlight intensity of the LCD screen.
  • Page 259 Appendix Parameter- Configuration File phone_setting.backlight_time <y0000000000xx>.cfg Configures the backlight time (in seconds) used to specify the delay time to turn off the backlight when the IP phone is inactive. Description If it is set to 300, the LCD backlight is turned off when the IP phone is inactive for 5 minutes.
  • Page 260 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File security.user_password <y0000000000xx>.cfg Configures a new administrator password for the IP phone. The IP phone uses “admin” as the default administrator password. Description A valid password should contain at least 6 characters, where at least one numeric and one alphabetic characters.
  • Page 261 Appendix and line keys are locked. All Keys: All keys are locked, except the Volume, Headset, Speakerphone and digit keys. Format Integer Default Value Valid values are: 0-All Keys Range 1-Function Keys 2-Menu Key Example phone_setting.phone_lock.lock_key_type = 2 Parameter- Configuration File <y0000000000xx>.cfg phone_setting.phone_lock.unlo ck_pin...
  • Page 262 Administrator’s Guide for SIP-T4X IP Phones Range 0 to 3600 Example phone_setting.phone_lock.lock_time_out = 8 NTP Server Parameter- Configuration File local_time.ntp_server1 <y0000000000xx>.cfg Configures the IP address or the domain name Description of the primary NTP server. Format IP Address or Domain Name Default Value cn.pool.ntp.org...
  • Page 263 Appendix Range 15 to 86400 Example local_time.interval = 1200 Time Zone Parameter- Configuration File local_time.time_zone <MAC>.cfg Defines the time zone. Description For more available time zone list, refer to Appendix B: Time Zones on page 231. Format Not Applicable Default Value Range -11 to +13 Example...
  • Page 264 Administrator’s Guide for SIP-T4X IP Phones Example local_time.summer_time = 2 Parameter- Configuration File local_time.dst_time_type <y0000000000xx>.cfg Configures the DST type. Note: It works only if the parameter Description “local_time.summer_time” is set to 1 (Enabled). Format Integer Default Value Valid values are:...
  • Page 265 Appendix of Day (For By Week) Default Value 1/1/0 1to 12/1 to 31/0 to 23 (for By Date) Range 1 to 12/1 to 5/1 to 7/0 to 23 (for By Week) Example local_time.start_time = 5/20/12 Parameter- Configuration File local_time.end_time <y0000000000xx>.cfg Specifies the time to end DST.
  • Page 266 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File local_time.offset_time <y0000000000xx>.cfg Configures the offset time (in minutes) of DST. Note: It works only when the parameter Description “local_time.summer_time” is set to 1 (Enabled). Format Integer Default Value Blank Range -300 to +300 Example local_time.offset_time = 120...
  • Page 267 Appendix 4-MM/DD/YY 5-DD MMM YYYY 6-WWW DD MMM Example local_time.date_format = 1 Parameter- Configuration File gui_lang.url <y0000000000xx>.cfg Specifies the access URL of the language pack. Note: The language packs you load are Description dependent on available language packs from the provisioning server. You can download the language pack to the phone user interface only.
  • Page 268 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File lang.gui <y0000000000xx>.cfg Specifies the language used on the phone Description user interface. Format String Default Value English Valid values are: English Chinese_S (not applicable to SIP-T42G/T41P) Chinese_T (not applicable to SIP-T42G/T41P)
  • Page 269 Appendix Portuguese Spanish Turkish Example lang.wui = French Parameter- Configuration File phone_setting.lcd_logo.mode <y0000000000xx>.cfg Configures the logo mode of the LCD screen. If it is set to 0 (Disabled), the IP phone is not allowed to display a logo. If it is set to 1 (System logo), the LCD screen will display the system logo.
  • Page 270 Administrator’s Guide for SIP-T4X IP Phones the custom logo file (logo.dob) from the provisioning server 192.168.10.25. lcd_logo.url = http://192.168.10.25/logo.dob Parameter- Configuration File features.pound_key.mode <y0000000000xx>.cfg Defines the "#" or "*" key as the send key. If it is set to 0 (Disabled), neither “#” nor “*”...
  • Page 271 Appendix Example features.send_key_tone = 0 Parameter- Configuration File 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 272 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File 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 it is set to 0 (Disabled), the Save Call Log option is hidden on the web user interface.
  • Page 273 Appendix If it is set to 0 (Disabled), there is no indicator displaying on the LCD screen, the IP phone does not log the missed call in the Missed Calls list. If it is set to 1 (Enabled), a prompt message "<number>...
  • Page 274 Administrator’s Guide for SIP-T4X IP Phones (Enabled). Format Integer Default Value Range 1 to 14 Example phone_setting.inter_digit_time = 1 Parameter- Configuration File call_waiting.enable <y0000000000xx>.cfg Enables or disables call waiting feature. If it is set to 0 (Disabled), a new incoming call...
  • Page 275 Appendix Configures the call waiting off code to Description deactivate the server-side call waiting feature. Format String Default Value Blank Range Not Applicable Example call_waiting.on_code = *56 Parameter- Configuration File call_waiting.tone <y0000000000xx>.cfg Enables or disables the playing of a call waiting tone when the IP phone receives an incoming call during a call.
  • Page 276 Administrator’s Guide for SIP-T4X IP Phones 1-Enabled Example auto_redial.enable = 1 Parameter- Configuration File auto_redial.interval <y0000000000xx>.cfg Configures the interval (in seconds) for the IP phone to wait between redials. Description The IP phone redials the dialed number at regular intervals till the callee answers the call.
  • Page 277 Appendix Note: The IP phone cannot automatically answer the incoming call during a call even if auto answer is enabled. Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.auto_answer = 1 Parameter- Configuration File features.auto_answer_delay <y0000000000xx>.cfg Configures the delay time (in seconds) of auto Description answer.
  • Page 278 Administrator’s Guide for SIP-T4X IP Phones Example features.call_completion_enable = 1 Parameter- Configuration File account.x.anonymous_call <MAC>.cfg Enables or disables anonymous call feature for account X. If it is set to 1 (Enabled), the IP phone blocks its identity from showing up to the callee when Description placing a call.
  • Page 279 Appendix Parameter- Configuration File <MAC>.cfg account.x.anonymous_call_onc Configures the anonymous call on code to activate the server-side anonymous call Description feature for account X (optional). X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example account.1.anonymous_call_oncode = *72 Parameter- Configuration File <MAC>.cfg...
  • Page 280 Administrator’s Guide for SIP-T4X IP Phones Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.reject_anonymous_call = 1 Parameter- Configuration File account.x.anonymous_reject_o <MAC>.cfg ncode Configures the anonymous call rejection on code to activate the server-side anonymous Description call rejection feature for account X (optional).
  • Page 281 Appendix Return Message When DND Parameter- Configuration File features.dnd_refuse_code <y0000000000xx>.cfg Defines 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 LCD screen. Description If it is set to 486 (Busy here), the caller’s LCD screen displays the reason “Busy here”...
  • Page 282 Administrator’s Guide for SIP-T4X IP Phones DND in Phone Mode Parameter- Configuration File features.dnd.enable <y0000000000xx>.cfg Enables or disables DND feature. Description If it is set to 1 (Enabled), the IP phone rejects incoming calls on all accounts. Format Boolean Default Value...
  • Page 283 Appendix server-side DND feature. Format String Default Value Blank Range Not Applicable Example features.dnd.off_code = *72 DND in Custom Mode Parameter- Configuration File account.x.dnd.enable <MAC>.cfg Enables or disables DND for account X. If it is set to 1 (Enabled), the IP phone rejects Description incoming calls on account X.
  • Page 284 Administrator’s Guide for SIP-T4X IP Phones X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example account.1.dnd.off_code = *74 Parameter- Configuration File features.busy_tone_delay <y0000000000xx>.cfg Configures a period of time (in seconds) for which the busy tone is audible on the IP phone.
  • Page 285 Appendix Format Integer Default Value Valid values are: 404-No Found Range 480-Temporarily not available 486-Busy here Example features.normal_refuse_code = 480 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.
  • Page 286 Administrator’s Guide for SIP-T4X IP Phones 1-Enabled Example sip.use_out_bound_in_dialog = 0 Parameter- Configuration File 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 287 Appendix Parameter- Configuration File account.x.advanced.timer_t4 <MAC>.cfg Configures the session timer of T4 (in seconds) for account X. T4 represents the time the network will take Description to clear messages between the SIP Client and SIP Server. X ranges from 1 to 6. Format Float Default Value...
  • Page 288 Administrator’s Guide for SIP-T4X IP Phones 1800 seconds. X ranges from 1 to 6. Format Integer Default Value 1800 Range 30 to 7200 Example account.1.session_timer.expires = 300 Parameter- Configuration File account.x.session_timer.refresher <MAC>.cfg Configures the session timer refresher for account X.
  • Page 289 Appendix Parameter- Configuration File features.play_hold_tone.delay <y0000000000xx>.cfg Specifies the interval (in seconds) at which the IP phone plays a hold tone. If it is set to 30, the IP phone plays a hold tone every 30 seconds when there is a hold call Description on the IP phone.
  • Page 290 Administrator’s Guide for SIP-T4X IP Phones Configures the call forward mode for the IP phone. If it is set to 0 (Phone), call forward feature is Description effective for the IP phone. If it is set to 1 (Custom), you can configure call forward feature for each account.
  • Page 291 Appendix Parameter- Configuration File forward.always.on_code < y0000000000xx >.cfg Configures the always forward on code to Description activate the server-side always forward feature. 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...
  • Page 292 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File forward.busy.target < y0000000000xx >.cfg Defines the destination number of the busy Description forward. Format String Default Value Blank Range Not Applicable Example forward.busy.target = 3602 Parameter- Configuration File forward.busy.on_code < y0000000000xx >.cfg...
  • Page 293 Appendix If it is set to 1 (Enabled), incoming calls are forward to the destination number after a period of ring time. Format Boolean Default Value 0-Disabled Range 1-Enabled Example forward.no_answer.enable = 1 Parameter- Configuration File forward.no_answer.target < y0000000000xx >.cfg Defines the destination number of the no Description answer forward.
  • Page 294 Administrator’s Guide for SIP-T4X IP Phones Default Value Blank Range Not Applicable Example forward.no_answer.on_code = *76 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...
  • Page 295 Appendix Format String Default Value Blank Range Not Applicable Example account.1.always_fwd.target = 3601 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...
  • Page 296 Administrator’s Guide for SIP-T4X IP Phones X ranges from 1 to 6. Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.busy_fwd.enable = 1 Parameter- Configuration File account.x.busy_fwd.target <MAC>.cfg Defines the destination number of the busy forward for account X. Description X ranges from 1 to 6.
  • Page 297 Appendix Format String Default Value Blank Range Not Applicable Example account.1.busy_fwd.off_code = *74 No Answer Forward Parameter- Configuration File account.x.timeout_fwd.enable <MAC>.cfg Enables or disables no answer forward feature for account X. If it is set to 1 (Enabled), incoming calls to the Description account X are forward to the destination number after a period of ring time.
  • Page 298 Administrator’s Guide for SIP-T4X IP Phones The interval of the ring time is n*6 (0≤n≤20), the valid values ranges from 0 to 20. X ranges from 1 to 6. Format Integer Default Value Range 0 to 20 Example account.1.timeout_fwd.timeout = 5...
  • Page 299 Appendix an incoming call to an international phone number (the prefix is 00). Format Boolean Default Value 0-Disabled Range 1-Enabled Example forward.international.enable = 1 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...
  • Page 300 Administrator’s Guide for SIP-T4X IP Phones Specifies whether to display the missed call Description prompt on the destination party’s phone. Format Boolean Default Value 0-Disabled Range 1-Enabled Example transfer.semi_attend_tran_enable = 1 Parameter- Configuration File account.x.conf_type <MAC>.cfg Defines the conference type for account X.
  • Page 301 Appendix account.1.conf_uri = Example conference@domain.com Parameter- Configuration File <y0000000000xx>.cfg transfer.tran_others_after_conf_e nable Enables or disables the phone to transfer call to the two parties after a local conference call hangup. If it is enabled, the other two parties remain Description connected when the conference initiator drops the conference call.
  • Page 302 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File <y0000000000xx>.cfg features.pickup.direct_pickup_co 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 303 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 <y0000000000xx>.cfg features.pickup.group_pickup_co Configures the group call pickup code on a phone basis. Description Note: The group call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
  • Page 304 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File account.x.dialoginfo_callpickup <MAC>.cfg Enables or disables the phone to pick up a call according to the SIP header of dialog-info for account X. Description If it is set to 1 (Enabled), call pickup is implemented through SIP signals.
  • Page 305 Appendix The default HTTP port is 80. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value Range 1 to 65535 Example network.port.http = 90 Parameter- Configuration File wui.https_enable <y0000000000xx>.cfg Enables or disables the IP phone to access its web user interface using HTTPS protocol.
  • Page 306 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File 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 307 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 308 Administrator’s Guide for SIP-T4X IP Phones 2-SIP INFO 3-AUTO or SIP INFO Example account.1.dtmf.type = 2 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 309 Appendix Parameter- Configuration File features.dtmf.hide <y0000000000xx>.cfg Enables or disables the IP phone to suppress the display of DTMF digits. Description If it is set to 1 (Enabled), the DTMF digits are displayed as asterisks. Format Boolean Default Value 0-Disabled Range 1-Enabled Example features.dtmf.hide = 1...
  • Page 310 Administrator’s Guide for SIP-T4X IP Phones performs the transfer as normal when pressing the transfer key during a call. If it is set to 1 (Enabled), the IP phone transmits the specified DTMF digits to the server for completing call transfer when pressing the transfer key during a call.
  • Page 311 Appendix call. Format Boolean Default Value 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 it is set to 0 (Disabled), the microphone is un-muted for incoming calls.
  • Page 312 Administrator’s Guide for SIP-T4X IP Phones 0-Disabled Range 1-Enabled Example features.intercom.tone = 1 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.
  • Page 313 Appendix Example features.alert_info_tone = 1 Parameter- Configuration File account.x.alert_info_url_enable <MAC>.cfg Enables or disables the distinctive ring tones by the Alert-Info SIP header for Description account X. X ranges from 1 to 6. Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.alert_info_url_enable = 1 Parameter- Configuration File...
  • Page 314 Administrator’s Guide for SIP-T4X IP Phones Valid values are: 1-Ring1.wav 2-Ring2.wav 3-Ring3.wav 4-Ring4.wav Range 5-Ring5.wav 6-Ring6.wav 7-Ring7.wav 8-Ring8.wav Ring 6-8 are not applicable to SIP-T42G/T41P . distinctive_ring_tones.alert_info.1.ringer Example Parameter- Configuration File voice.tone.country <y0000000000xx>.cfg Description Configures the tone type for the IP phone.
  • Page 315 Appendix Mexico  New Zealand  Netherlands  Norway  Portugal  Spain  Switzerland  Sweden  Russia  United States  Chile  Czech ETSI  Example voice.tone.country = Austria Parameter- Configuration File voice.tone.dial <y0000000000xx>.cfg voice.tone.ring voice.tone.busy voice.tone.congestion voice.tone.callwaiting voice.tone.dialrecall voice.tone.record...
  • Page 316 Administrator’s Guide for SIP-T4X IP Phones 200+300/500, 600+700+800+1000/2000). The exclamation point (!) can be added optionally, which means these tones are only played once. Note: It works only if the parameter “voice.tone.country” is set to Custom. Format Refer to the introduction above...
  • Page 317 Appendix Parameter- Configuration File <y0000000000xx>.cfg features.remote_phonebook.flas h_time Specifies how often to refresh the local cache of the remote phone book. Description If it is set to 3600, the IP phone refreshes the local cache of the remote phone book every 3600 seconds.
  • Page 318 Administrator’s Guide for SIP-T4X IP Phones Range Not Applicable ldap.name_filter = (|(cn=%)(sn=%)) When the name prefix of the cn or sn of the Example contact record matches the search criteria, the record will be displayed on the LCD screen. Parameter- Configuration File ldap.number_filter...
  • Page 319 Format String Default Value Blank Range Not Applicable Example ldap.base = dc=yealink,dc=cn Parameter- Configuration File ldap.user <y0000000000xx>.cfg Specifies 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 320 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File ldap.password <y0000000000xx>.cfg Specifies the password to login the LDAP server. This parameter can be left blank in case the Description server allows anonymous to login. Otherwise you will need to provide the password to access the LDAP server.
  • Page 321 Appendix Default Value Blank Range Not Applicable Example ldap.name_attr = cn sn Parameter- Configuration File ldap.numb_attr <y0000000000xx>.cfg Specifies the number attributes of each record to be returned by the LDAP server. It Description compresses the search results. You can configure multiple number attributes separated by space.
  • Page 322 Administrator’s Guide for SIP-T4X IP Phones Default Value Range 2 or 3 Example ldap.version = 3 Parameter- Configuration File ldap.call_in_lookup <y0000000000xx>.cfg Enables or disables the IP phone to perform Description an LDAP search when receiving an incoming call. Format Boolean...
  • Page 323 Appendix receives an incoming call. Format Boolean Default Value 0-Disabled Range 1-Enabled Example features.pickup.blf_visual_enable = 1 Parameter- Configuration File <y0000000000xx>.cfg features.pickup.blf_audio_enabl Enables or disables the IP phone to play an Description alert tone when the monitored user receives an incoming call. Format Boolean Default Value...
  • Page 324 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File account.x.music_server_uri <MAC>.cfg Specifies the Music on Hold server address. Examples for valid values: <10.1.3.165>, 10.1.3.165, sip:moh@ucap.com, <sip:moh@ucap.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 325 Appendix available. Format Boolean Default Value 0- Disabled Value 1- Enabled Example acd.auto_available = 1 Parameter- Configuration File acd.auto_available_timer <y0000000000xx>.cfg Specifies the length of time (in seconds) Description before the IP phone state is automatically reset to “available”. Format Integer Default Value Value 0 to 120...
  • Page 326 Administrator’s Guide for SIP-T4X 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 327 Appendix Format String Default Value Black Value Not Applicable Example voice_mail.number.1 = 3606 Parameter- Configuration File multicast.codec <y0000000000xx>.cfg Specifies a multicast codec for the IP phone Description to use to send an RTP stream. Format string Default Value G722 Valid values are: PCMU ...
  • Page 328 Administrator’s Guide for SIP-T4X IP Phones Format Boolean Default Value 0-Disabled Range 1-Enabled Example multicast.receive_priority.enable =1 Parameter- Configuration File 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 329 Appendix 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 Parameter- Configuration File action_url.setup_completed = <y0000000000xx>.cfg action_url.log_on = action_url.log_off = action_url.register_failed = action_url.off_hook = action_url.on_hook =...
  • Page 330 Administrator’s Guide for SIP-T4X IP Phones 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 = Specifies the URL for the predefined event. The value format is: http(s)://IP address of server/help.xml? variable name=variable...
  • Page 331 Appendix Parameter- Configuration File features.action_uri_limit_ip <y0000000000xx>.cfg Specifies 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 332 Administrator’s Guide for SIP-T4X IP Phones sip:user@example.com account.1.sip_server.2.address = 192.168.1.15 Parameter- Configuration File account.x.sip_server.y.port <MAC>.cfg Configures the SIP server port. X ranges from 1 to 6. Description Y ranges from 1 to 2. Format Integer Default Value 5060 Range 0 to 65535 Example account.1.sip_server.1.port = 5060...
  • Page 333 Appendix Range 0 to 20 Example account.1.sip_server.1.retry_counts = 3 Fallback Mode Parameter- Configuration File <MAC>.cfg account.x.fallback.redundancy_ty 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: 0-Concurrent registration: the phone registers to the working server and fallback server at the same time.

  • Page 334: Failover Mode

    Administrator’s Guide for SIP-T4X IP Phones Failover Mode Parameter- Configuration File <MAC>.cfg account.x.sip_server.y.failback_mo Configures the way in which the phone fails back to the primary server for call control when in the failover mode. Description X ranges from 1 to 6.
  • Page 335 Appendix When the value is configured between 1 and 59, the phone automatically sets the time interval to be 60. Note: This parameter is only valid when the parameter “account.x.sip_server.y.failback_mode” is configured to 3. X ranges from 1 to 6. Y ranges from 1 to 2.
  • Page 336 Administrator’s Guide for SIP-T4X IP Phones port is given, the IP phone performs the DNS NAPTR and SRV queries for the transport protocol, port and IP address. X ranges from 1 to 6. Format Integer Default Value Range Valid values are:...
  • Page 337 Appendix Default Value 0-Disabled Range 1-Enabled Example network.lldp.enable = 1 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.
  • Page 338 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File network.vlan.internet_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.
  • Page 339 Appendix 1-Enabled Example network.vlan.pc_port_enable = 1 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...
  • Page 340 Administrator’s Guide for SIP-T4X IP Phones Format Boolean Default Value 0-Disabled Range 1-Enabled Example network.vlan.dhcp_enable = 1 Parameter- Configuration File network.vlan.dhcp_option <y0000000000xx>.cfg Specifies the DHCP option from which the phone will obtain the VLAN settings. Description You can configure at most five DHCP options, and separate options by comma.
  • Page 341 Appendix Specifies the access URL of the OpenVPN Description tar package. Format String Default Value Blank Range Not Applicable openvpn.url = Example http://192.168.10.25/OpenVPN.tar Parameter- Configuration File network.qos.rtptos <y0000000000xx>.cfg Configures the DSCP for voice packets. The default DSCP value for RTP packets is 46 (Expedited Forwarding).
  • Page 342 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File account.x.nat.nat_traversal <MAC>.cfg Enables or disables the NAT traversal for account X. Description X ranges from 1 to 6. Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.nat.nat_traversal = 1 Parameter- Configuration File account.x.nat.stun_server...
  • Page 343 Appendix Parameter- Configuration File network.snmp.enable <y0000000000xx>.cfg Enables or disables SNMP feature on the IP phone. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Boolean Default Value 0-Disabled Range 1-Enabled Example network.snmp.enable = 0 Parameter-...
  • Page 344 Administrator’s Guide for SIP-T4X IP Phones can accept and handle GET requests from any IP address. If the value is left blank, the IP phone cannot receive or handle any GET request. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 345 Appendix phone will reboot to make the change take effect. Format String Default Value Blank Range Not Applicable Example network.802_1x.identity = admin Parameter- Configuration File network.802_1x.md5_password <y0000000000xx>.cfg Enters the password used for authenticating the IP phone. Note: If you change this parameter, the IP Description phone will reboot to make the change take effect.
  • Page 346 Administrator’s Guide for SIP-T4X IP Phones network.802_1x.root_cert_url = Example http://192.168.1.10/ca.pem Parameter- Configuration File network.802_1x.client_cert_url <y0000000000xx>.cfg Specifies the access URL of the client certificate used for authentication. Note: If you change this parameter, the IP Description phone will reboot to make the change take effect.
  • Page 347 Appendix 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.username = user1 Parameter- Configuration File...
  • Page 348 Administrator’s Guide for SIP-T4X IP Phones Configures the user name 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. Format String...
  • Page 349 Appendix Default Value 0-Disabled Range 1-Enabled managementserver.periodic_inform_enable = Example Parameter- Configuration File managementserver.periodic_in <y0000000000xx>.cfg form_interval Configures the interval (in seconds) to report its configuration information to the ACS. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 350 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File network.ipv6_internet_port.type <MAC>.cfg Specifies the IPv6 address assignment method. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value Valid values are:...
  • Page 351 Appendix Parameter- Configuration File <MAC>.cfg network.ipv6_internet_port.gat eway Configures the gateway when the Internet port type is defined as Static IP Address. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank...
  • Page 352 Administrator’s Guide for SIP-T4X IP Phones Format IP Address Default Value Blank Range Not Applicable network.ipv6_secondary_dns = Example 2026:1234:1:1:c3c7:c11c:5447:23a6 Parameter- Configuration File 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 the ICMPv6 protocol.
  • Page 353 Appendix 0-Disabled Range 1-Enabled Example features.headset_prior = 1 Parameter- Configuration File features.headset_training <y0000000000xx>.cfg Enables or disables dual headset feature. If it is 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 354 Administrator’s Guide for SIP-T4X IP Phones When Y=6, the default value is 1; When Y=7, the default value is 0; When Y=8, the default value is 0; When Y=9, the default value is 0; When Y=10, the default value is 0;...
  • Page 355 Appendix Valid values are: PCMU  PCMA  G729  G722  G723_53  Range G723_63  G726_16  G726_24  G726_32  G726_40  iLBC   account.1.codec.1.payload_type = Example G723_53 Parameter- Configuration File account.x.codec.y.priority <MAC>.cfg Specifies the priority for the codec. Description X ranges from 1 to 6.
  • Page 356 Administrator’s Guide for SIP-T4X IP Phones Range 0 to 13 Example account.1.codec.1.priority = 1 Parameter- Configuration File account.x.codec.y.rtpmap <MAC>.cfg Configures the rtpmap. Description X ranges from 1 to 6. Y ranges from 1 to 14. Format Integer When Y=1, the default value is 0;...
  • Page 357 Appendix Configures the ptime (in milliseconds) for the codec. Description X ranges from 1 to 6. Format Integer Default Value Valid values are: Range 0 (Disabled) 10, 20, 30, 40, 50, 60 Example account.1.ptime = 30 Parameter- Configuration File voice.echo_cancellation <y0000000000xx>.cfg Enables or disables AEC feature on the IP Description...
  • Page 358 Administrator’s Guide for SIP-T4X IP Phones Parameter- Configuration File voice.cng <y0000000000xx>.cfg Enables or disables CNG feature on the IP Description phone. Format Boolean Default Value 0-Disabled Range 1-Enabled Example voice.cng = 1 Parameter- Configuration File voice.jib.adaptive <y0000000000xx>.cfg Description Configures the type of jitter buffer.
  • Page 359 Appendix Parameter- Configuration File voice.jib.max <y0000000000xx>.cfg Configures the maximum 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 60 to 300 Example voice.jib.max = 200 Parameter- Configuration File voice.jib.normal...
  • Page 360 Administrator’s Guide for SIP-T4X IP Phones Default Value 0 (UDP) Valid values are: 0-UDP Range 1-TCP 2-TLS 3-DNS-NAPTR Example account.1.transport = 2 Parameter- Configuration File security.trust_certificates <y0000000000xx>.cfg Enables or disables the IP phone to authenticate the connecting server. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 361 Appendix Parameter- Configuration File security.cn_validation <y0000000000xx>.cfg Enables or disables the IP phone to mandatorily validate the CommonName or subjectAltName of the certificate sent by the connecting server. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
  • Page 362 Administrator’s Guide for SIP-T4X IP Phones Format String Default Value Blank Range Not Applicable trusted_certificates.url = Example http://192.168.1.20/tc.crt Parameter- Configuration File server_certificates.url <y0000000000xx>.cfg Specifies the access URL of the certificate the IP phone sends for authentication. Description Note: The certificate you want to upload must be in .pem or .cer format.
  • Page 363 Appendix Example account.1.srtp_encryption = 1 Parameter- Configuration File auto_provision.aes_key_in_file <y0000000000xx>.cfg Enable or disable the IP phone to decrypt configuration files using the encypted AES key. If it is set to 1 (Enabled), the IP phone will download <y0000000000xx_Security>.enc and <MAC_Security>.enc files during auto Description provisioning, and then decrypts these files into the plaintext keys (e.g., key2, key3)
  • Page 364 Administrator’s Guide for SIP-T4X IP Phones auto_provision.aes_key_16.com = Example 0123456789abcdef Parameter- Configuration File auto_provision.aes_key_16.mac <y0000000000xx>.cfg Configures the plaintext AES key which is used to decrypt the <MAC>.cfg file. Description Note: It works only if the parameter “auto_provision.aes_key_in_file” is set to 0 (Disabled).
  • Page 365 Appendix provisioning server address during startup. Format Boolean Default Value 0-Disabled Range 1-Enabled Example auto_provision.pnp_enable = 0 Parameter- Configuration File auto_provision.repeat.enable < y0000000000xx >.cfg Enables or disables the phone to check new Description configuration repeatedly. Format Boolean Default Value 0-Disabled Range 1-Enabled Example...
  • Page 366 Administrator’s Guide for SIP-T4X IP Phones firmware.url = Example http://192.168.1.20/2.70.0.50.rom Parameter- Configuration File dialplan_replace_rule.url <y0000000000xx>.cfg Specifies the access URL of the replace rule Description template. Format Default Value Blank Range Not Applicable dialplan_replace_rule.url = Example http://192.168.10.25/dialplan.xml Parameter- Configuration File dialplan_dialnow.url <y0000000000xx>.cfg...
  • Page 367 Appendix screen when in the CallFailed state. Format Default Value Not Applicable Range Not Applicable The following example uses HTTP to download the CallFailed state file from the “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. Example custom_softkey_call_failed.url = http://10.2.8.16:8080/XMLfiles/CallFailed.xm Parameter- Configuration File...
  • Page 368 Administrator’s Guide for SIP-T4X IP Phones download the Connecting state file from the “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_connecting.url = http://10.2.8.16:8080/XMLfiles/Connecting.x Parameter- Configuration File custom_softkey_dialing.url <y0000000000xx>.cfg Specifies the access URL of the customized Description file for the soft key presented on the LCD screen when in the Dialing state.
  • Page 369 Appendix Parameter- Configuration File custom_softkey_talking.url <y0000000000xx>.cfg Specifies the access URL of the customized Description file for the soft key presented on the LCD screen when in the Talking state. Format Default Value Not Applicable Range Not Applicable The following example uses HTTP to download the Talking state file from the “XMLfiles”...
  • Page 370 Administrator’s Guide for SIP-T4X IP Phones Default Value Blank Range Not Applicable remote_phonebook.data.1.url = Example http://192.168.1.20/phonebook.xml Parameter- Configuration File wallpaper_upload.url <y0000000000xx>.cfg Specifies the access URL of the wallpaper image. Description Note: It is only applicable to the SIP-T46G IP phone.
  • Page 371 Appendix Parameter- Configuration File syslog.server <y0000000000xx>.cfg Specifies the IP address of the syslog server where to export the log files. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank Range...
  • Page 372 Administrator’s Guide for SIP-T4X IP Phones This section provides DSS key parameters you can configure on the IP phone. DSS key can be assigned with various key features. The parameters of the DSS key are detailed in the following: Parameter- Configuration File linekey.x.type...
  • Page 373 Appendix Zero Touch   Local Group  Keypad Lock  Directory  Format Integer SIP-T46G: 15 for line key 1-6, 0 for line key 7-27. Default Value SIP-T42G/T41P: 15 for line key 1-3, 0 for line key 4-15. Valid values are: 0-N/A 1-Conference 2-Forward...
  • Page 374 Administrator’s Guide for SIP-T4X IP Phones Example linekey.1.type = 61 Parameter- Configuration File linekey.x.line <y0000000000xx>.cfg Specifies the desired line to apply the key feature. When Local Group or XML Group is assigned to the line key, this parameter is used to specify the desired phonebook (or group) if multiple phonebooks (or groups) are configured on the IP phone.
  • Page 375 Appendix For local group and XML group, valid values are: 0 stands for the first phonebook (or group), 1 stands for the second phonebook (or group) and so on. Example linekey.1.line = 2 Parameter- Configuration File linekey.x.value <y0000000000xx>.cfg Specifies the value for some key features. Description X ranges from 1 to 27 (for SIP-T46G).
  • Page 376 Administrator’s Guide for SIP-T4X IP Phones This is an optional configuration. X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). String Format Default Value Blank Range Not Applicable Example linekey.1.label = Dir Keypad Lock Key...
  • Page 377 Appendix Parameter- Configuration File linekey.x.type <y0000000000xx>.cfg Configures a line key to be directed call pickup key on the IP phone. The digit 9 stands for the key type Directed Description Pickup. X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). Format Integer Value...
  • Page 378 Administrator’s Guide for SIP-T4X IP Phones Group Call Pickup Key Parameter- Configuration File linekey.x.type <y0000000000xx>.cfg Configures a line key to be group call pickup key on the IP phone. The digit 23 stands for the key type Group Description Pickup.
  • Page 379 Appendix Call Return Key Parameter- Configuration File linekey.x.type <y0000000000xx>.cfg Configures a line key to be call return key on the IP phone. The digit 7 stands for the key type Call Description Return. X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P).
  • Page 380 Administrator’s Guide for SIP-T4X IP Phones Call Park Key Parameter- Configuration File linekey.x.type <y0000000000xx>.cfg Configures a line key to be call park key on the IP phone. The digit 10 stands for the key type Call Description Park. X ranges from 1 to 27 (for SIP-T46G).

  • Page 381: Intercom Key

    Appendix Intercom Key Parameter- Configuration File linekey.x.type <y0000000000xx>.cfg Configures a line key to be the intercom key. The digit 14 stands for the key type Description Intercom. X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). Format Integer Value...
  • Page 382 Administrator’s Guide for SIP-T4X IP Phones LDAP Key Parameter- Configuration File linekey.x.type <y0000000000xx>.cfg Configures a line key to be LDAP key on the IP phone. The digit 38 stands for the key type LDAP. Description X ranges from 1 to 27 (for SIP-T46G).
  • Page 383 Appendix Example linekey.3.line = 2 Parameter- Configuration File linekey.x.value <y0000000000xx>.cfg Specifies the number of the monitored user. Description X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). Format String Range Not Applicable Example linekey.3.value = 1008 Parameter- Configuration File...
  • Page 384 Administrator’s Guide for SIP-T4X IP Phones Example linekey.2.type = 42 Multicast Paging Key Parameter- Configuration File linekey.x.type <y0000000000xx>.cfg Configures a line key to be a multicast paging key on the IP phone. The digit 24 stands for the key type Description Multicast Paging.
  • Page 385 Appendix X ranges from 1 to 15 (for SIP-T42G/T41P). Format Integer Value Example linekey.2.type = 25 URL Record Key Parameter- Configuration File linekey.x.type <y0000000000xx>.cfg Configures a line key to be a URL record key on the IP phone. The digit 35 stands for the key type URL Description Record.
  • Page 386 Administrator’s Guide for SIP-T4X IP Phones key on the IP phone. The digit 34 stands for the key type Hot Desking. X ranges from 1 to 27 (for SIP-T46G). X ranges from 1 to 15 (for SIP-T42G/T41P). Format Integer Value Example linekey.2.type = 34...
  • Page 387 Appendix This section describes how Yealink SIP-T4X IP phones comply with the IETF definition of SIP as described in RFC 3261. This section contains compliance information in the following: RFC and Internet Draft Support  SIP Request  SIP Header ...
  • Page 388 Administrator’s Guide for SIP-T4X IP Phones RFC 3891—The Session Initiation Protocol (SIP) “Replaces” Header  RFC 3892—The Session Initiation Protocol (SIP) Referred-By Mechanism  RFC 3968—The Internet Assigned Number Authority (IANA) Header Field  Parameter Registry for the Session Initiation Protocol (SIP) RFC 3969—The Internet Assigned Number Authority (IANA) Uniform Resource...
  • Page 389 Appendix Method Supported Notes CANCEL OPTIONS SUBSCRIBE NOTIFY REFER PRACK INFO MESSAGE UPDATE PUBLISH 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...
  • Page 390 Administrator’s Guide for SIP-T4X IP Phones Notes Method Supported Expires From Max-Forwards Min-SE P-Asserted-Identity P-Preferred-Identity Proxy-Authenticate Proxy-Authorization RAck Record-Route Refer-To Referred-By Remote-Party-ID Replaces Require Route RSeq Session-Expires Subscription-State Supported User-Agent The following SIP responses are supported: 1xx Response—Information Responses 1xx Response...
  • Page 391 Appendix 1xx Response Supported Notes 180 Ringing 181 Call Is Being Forwarded 183 Session Progress 2xx Response—Successful Responses 2xx Response Supported Notes 200 OK In REFER transfer. 202 Accepted 3xx Response—Redirection Responses 3xx Response Supported Notes 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 4xx Response—Request Failure Responses 4xx Response...
  • Page 392 Administrator’s Guide for SIP-T4X IP Phones 4xx Response Supported Notes 413 Request Entity Too Large 414 Request-URI Too Long 415 Unsupported Media Type 416 Unsupported URI Scheme 420 Bad Extension 421 Extension Required 423 Interval Too Brief 480 Temporarily Unavailable...
  • Page 393 Appendix 6xx Response—Global Responses 6xx Response Supported Notes 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable SDP Headers Supported v—Protocol version o—Owner/creator and session identifier a—Media attribute c—Connection information m—Media name and transport address s—Session name t—Active time...
  • Page 394 Administrator’s Guide for SIP-T4X IP Phones SIP uses six request methods: INVITE—Indicates a user is being invited to participate in a call session.  ACK—Confirms that the client has received a final response to an INVITE request.  BYE—Terminates a call and can be sent by either the caller or the callee.
  • Page 395 Appendix 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 396 Administrator’s Guide for SIP-T4X 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 397 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 398 Administrator’s Guide for SIP-T4X 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 399 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 400 Administrator’s Guide for SIP-T4X 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 401 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 402 Administrator’s Guide for SIP-T4X IP Phones Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
  • Page 403 Appendix Step Action Description User B User A wants to disconnect the call. User B sends a SIP 200 OK response to 200 OK—User B to Proxy the proxy server. The SIP 200 OK Server response indicates that User B has received the CANCEL request.
  • Page 404 Administrator’s Guide for SIP-T4X IP Phones 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 405 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 406 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...
  • Page 407 Appendix network. The call flow scenario is as follows: User A calls User B. User B answers the call. User C calls User B. User B accepts the call from User C. Proxy Server User C User A User B F1.
  • Page 408 Administrator’s Guide for SIP-T4X IP Phones Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted ...
  • Page 409 Appendix Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to User B.
  • Page 410 Administrator’s Guide for SIP-T4X IP Phones Step Action Description User A sends a mid-call INVITE request INVITE—User A to Proxy to the proxy server with new SDP Server session parameters, which are used to place the call on hold. INVITE—Proxy Server to User The proxy server forwards the mid-call INVITE message to User B.
  • Page 411 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 412 Administrator’s Guide for SIP-T4X 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 413 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 414 Administrator’s Guide for SIP-T4X IP Phones Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active.
  • Page 415 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 416 Administrator’s Guide for SIP-T4X IP Phones User A transfers the call to User C. Call is established between User B and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4.
  • Page 417 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 418 Administrator’s Guide for SIP-T4X IP Phones Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active.
  • Page 419 Appendix Step Action Description sends the INVITE request to User C. User C sends a SIP 180 Ringing 180 Ringing—User C to Proxy response to the proxy server. The 180 Server Ringing response indicates that the user is being alerted. The proxy server forwards the 180 180 Ringing—Proxy Server to Ringing response to User A.
  • Page 420 Administrator’s Guide for SIP-T4X 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 421 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 422 Administrator’s Guide for SIP-T4X IP Phones Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of the User B is ...
  • Page 423 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 unique Call-ID is generated and the Server Contact-URI field indicates that User A requested the call. The proxy server maps the SIP URI in the INVITE—Proxy Server to User To field to User C.
  • Page 424 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 425 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 426 Administrator’s Guide for SIP-T4X IP Phones Step Action Description ACK 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 427 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 428 Administrator’s Guide for SIP-T4X 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 429 Appendix Step Action Description ACK 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 430 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 431 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 432 Administrator’s Guide for SIP-T4X 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 433 Appendix Step Action Description User A sends a SIP ACK to the proxy server. The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to User B.
  • Page 434 Administrator’s Guide for SIP-T4X IP Phones Step Action Description sends the SIP INVITE request to User C. User C sends a SIP 180 Ringing 180 Ringing—User C to Proxy response to the proxy server. The 180 Server Ringing response indicates that the user is being alerted.

  • Page 435: Network Settings

    Appendix 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 436: Time Settings

    Administrator’s Guide for SIP-T4X IP Phones dialplan.replace.prefix.X = dialplan.replace.replace.X = dialplan.replace.line_id.X = 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 =...

  • Page 437: Call Forward

    Appendix sip.rfc2543_hold = #Hotline features.hotline_number = features.hotline_delay = #Web Server Type wui.http_enable = network.port.http = wui.https_enable = 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 =...

  • Page 438: Call Transfer

    Administrator’s Guide for SIP-T4X IP Phones account.1.timeout_fwd.target = account.1.timeout_fwd.timeout = account.1.timeout_fwd.on_code = 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 =...

  • Page 439: Action Url

    Appendix ldap.number_filter = ldap.host = 0.0.0.0 ldap.port = 389 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 =...
  • Page 440 Administrator’s Guide for SIP-T4X IP Phones 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 = #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 =...
  • Page 441 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 Call Waiting Tone Always Forward Calling Line Identification Presentation Analyzing the Configuration Files...
  • Page 442 Administrator’s Guide for SIP-T4X IP Phones Getting Information from Status Indicators NAT Traversal Getting Started Network Address Translation (NAT) Group Call Pickup Network Conference No Answer Forward H.323 Headset Prior Phone Lock Hotline Phone User Interface Hot Desking Physical Features of SIP-T4X IP Phones...
  • Page 443 Index SRTP STUN Server Suppressing DTMF Display 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 User Agent Client (UAC) User Agent Server (UAS) User Password...

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