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Note: This device is tested and complies with the limits for a Class B digital device, pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications.
IP phone’s performance in the network. So an understanding of IP networking and prior knowledge of IP telephony concepts are necessary. This guide covers the SIP-T38G and T32G IP phones. The following related documents for the SIP-T3xG IP phones are available: Quick Installation Guides, which describe how to assemble the IP phones.
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Administrator’s Guide for SIP-T3xG IP Phones Chapter 2, ―Getting Started‖ describes how to install and connect the IP phones and the IP phone interface methods. Chapter 3, ―Configuring Basic Features‖ describes how to configure the basic features on the IP phones. Chapter 4, ―Configuring Advanced Features‖...
Table of Contents About This Guide ..............v Documentations ..........................v In This Guide ............................ v Table of Contents ..............vii Product Overview ..............1 VoIP Principle ............................ 1 SIP Components..........................2 SIP IP Phone Models ........................3 Physical Features of the SIP-T3xG IP Phones ................4 Key Features of the SIP-T3xG IP Phones ..................
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Administrator’s Guide for SIP-T3xG IP Phones Backlight ............................39 User Password ..........................41 Administrator Password ........................ 42 Phone Lock ............................. 44 Time and Date ..........................46 Language ............................53 Loading Language Packs ...................... 53 Specifying the Language to Use................... 54 Softkey Layout..........................
Product Overview This chapter contains the following information about the SIP-T3xG 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.
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Administrator’s Guide for SIP-T3xG IP Phones SIP provides the 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 ...
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The SIP-T3xG IP phones also support advanced functionalities, including LDAP , Busy Lamp Field, Shared Call Appearance and Network Conference. The following IP phone models are described: SIP-T38G SIP-T32G ...
A call server is active and configured to receive and send SIP messages. This section lists the available physical features of the SIP-T3xG IP phones. SIP-T38G Physical Features: TI Aries chipset and TI voice engine 4.3‖ TFT-LCD, 480 x 272 pixel, 16.7M colors...
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Product Overview SIP-T32G Physical Features: TI Aries chipset and TI voice engine 3‖ TFT-LCD, 400 x 240 pixel, 262K colors 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated HD Voice: HD Codec, HD Handset, HD Speaker 32 keys including 3 programmable keys 1xRJ9 (4P4C) handset port 1xRJ9 (4P4C) headset port 2xRJ45 10/100/1000Mbps Ethernet ports 5 LEDs: 1xpower, 3xline, 1xmessage...
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Administrator’s Guide for SIP-T3xG IP Phones Advanced Features: BLF/BLF list, shared call appearance, distinctive ring tones, remote phonebook, LDAP , 802.1x authentication. Codecs and Voice Features Wideband codec: G.722 Narrowband codec: G.711, G.723.1, G.726, G.729AB VAD, CNG, AEC, PLC, AJB, AGC Full-duplex speakerphone with AEC Network Features ...
Getting Started This chapter introduces the initialization of the SIP-T3xG IP phones, the installing and connecting process of the IP phones which you need to follow. This chapter provides the following major sections: Connecting the IP Phones Initialization Process Overview ...
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Administrator’s Guide for SIP-T3xG IP Phones Attach the stand: SIP-T38G Desk-mount Method Wall-mount Method SIP-T32G...
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Getting Started Connect the handset and optional headset: SIP-T38G SIP-T32G Connect the network and power: AC power Power over Ethernet (PoE) AC Power To connect the AC power and network: Connect the DC plug of the power adapter to the DC5V port on the IP phone and...
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Administrator’s Guide for SIP-T3xG IP Phones Connect the supplied Ethernet cable between the Internet port on the IP phone and the Internet port in your network or switch/hub device port. Power over Ethernet Using a regular Ethernet cable, the IP phones can be powered from a PoE (IEEE 802.3af) compliant switch or hub.
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Getting Started The initialization process of the IP phone is responsible for network connectivity and operation of the IP phone in your local network. Once you connect your IP phone to the network and to an electrical supply, the IP phone begins its initialization process.
Administrator’s Guide for SIP-T3xG IP Phones downloaded firmware file differs from that of the image stored in the flash memory, the IP phone performs a firmware update. Downloading the resource files In addition to configuration file(s), the IP phone may require resource files before it can deliver service.
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The Common CFG file has a fixed name for each IP phone model, while the MAC-Oriented CFG file is named after the MAC address of the IP phone. For example, if the MAC address of the SIP-T38G IP phone is 001565113af8, the names of these two configuration files must be: y000000000038.cfg and 001565113af8.cfg.
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Administrator’s Guide for SIP-T3xG IP Phones When modifying parameters, remember the following: Parameters in the configuration files override those stored in the IP phone’s flash memory. The .cfg extension of the configuration files must be in lowercase. Each line in a configuration file must use the following format and adhere to the ...
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Getting Started T38G/T32G Description Headset mode Multi-lingual lowercase letters input mode Multi-lingual uppercase letters input mode Alphanumeric input mode Numeric input mode Voice Mail Text Message Auto Answer Do Not Disturb Call Forward Call Hold Call Mute Ringer volume is 0 Phone Lock Missed Calls Received Calls...
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Administrator’s Guide for SIP-T3xG IP Phones T38G/T32G Description Recording box is full A call cannot be recorded Recording starts successfully Recording cannot be started Recording cannot be stopped Open VPN Conference The default contact photo This section describes how to configure the basic network parameters that are required for the IP phones to operate in the network.
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Getting Started When the IP phones are simply plugged into the network, the DHCP process begins. The IP phones broadcast DISCOVER messages to request the network information carried in DHCP options and the DHCP server responds with the specific values in the corresponding options.
Administrator’s Guide for SIP-T3xG IP Phones Procedure DHCP can be configured using the configuration files or locally. Configure DHCP on the IP phone. Configuration File <y0000000000xx>.cfg For more information, refer to DHCP on page 224. Configure DHCP on the IP phone. Navigate to: Web User Interface http://<phoneIPAddress>/cgi-bin/...
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Getting Started To configure DHCP via phone user interface: Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN Port. Press to highlight the DHCP IP Client field, and then press the Enter soft key. The IP phone reboots automatically to make the settings effective after a period of time.
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Administrator’s Guide for SIP-T3xG IP Phones Enter the IP address, subnet mask, default gateway, primary DNS and secondary DNS in the corresponding fields. Click Confirm to accept the change. The web user interface pops up a dialog box to prompt that the settings will take effect after reboot.
Getting Started PPPoE (Point-to-Point Protocol over Ethernet) is a network protocol that is used by Internet Service Providers (ISPs) to provision 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.
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Administrator’s Guide for SIP-T3xG IP Phones Enter the username and password in the corresponding fields. Click Confirm to accept the change. The web user interface pops up a dialog box to prompt that the settings will take effect after reboot. Click OK to reboot the IP phone.
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Getting Started The PC port is used to connect a PC behind the IP phone, which can be configured as the following two modes: Bridge: In the bridge mode, the IP phone is considered as a bridge, the PC attached to the PC port appears on the network as a stand-alone device with its own IP address.
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Administrator’s Guide for SIP-T3xG IP Phones 5) (If enabled) Enter the end IP address in the End IP Address field. Click Confirm to accept the change. The web user interface pops up a dialog box to prompt that the settings will take effect after reboot.
Getting Started Regular expression, often called a pattern, is an expression that specifies a set of strings. A regular expression provides a concise and flexible means to "match" (specify and recognize) strings of text, such as particular characters, words, or patterns of characters. Regular expression is used by many text editors, utilities, and programming languages to search and manipulate text based on patterns.
Administrator’s Guide for SIP-T3xG IP Phones stands for the corresponding parenthesis. Example: A replace rule configuration, Prefix: "001(xxx)45(xx)", Replace: "9001$145$2". When you dial out "0012354599" on your phone, the IP phone will replace the number with "90012354599". ―$1‖ means 3 digits in the first parenthesis, that is, ―235‖.
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Getting Started If you leave the field blank, the replace rule applies to all accounts on the IP phone. Click Add to add the replace rule. Click Save to accept the change. Dial-now is a string that is used to match the numbers entered by the user. When entered numbers match the predefined dial-now rule, the IP phone will automatically dial out the numbers without pressing the send key.
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Administrator’s Guide for SIP-T3xG IP Phones Plan on page 231. Create the dial-now rule for the IP phone. Navigate to: http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Phone-Dial Now.htm Local Web User Interface Configure the delay time for the dial-now rule. Navigate to: http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Phone-Feat ures.htm To create the dial-now rule via web user interface: Click on Phone->Dial Plan->Dial-now.
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Getting Started Enter the desired time in the Dial-now Time-out (seconds) field. Click Confirm to accept the change. Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country. When entered numbers match the predefined area code rule, the IP phone will automatically add the area code to the beginning of the dialed numbers.
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Administrator’s Guide for SIP-T3xG IP Phones http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Phone-Are aCode.htm To configure an area code rule via web user interface: Click on Phone->Dial Plan->Area Code. Enter the desired values in the Code, Minimum Length (1-15) and Maximum Length (1-15) fields. Enter the desired line ID in the Account field or leave it blank. If you leave the field blank, the area code rule applies to all accounts on the IP phone.
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Getting Started Create the block out rule for the desired line. Navigate to: Local Web User Interface http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Phone-Bloc kOut.htm To create the block out rule via web user interface: Click on Phone->Dial Plan->Block Out. Enter the desired value in the Block Out Number field. Enter the desired line ID in the Account field or leave it blank.
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Administrator’s Guide for SIP-T3xG IP Phones...
Configuring Basic Features This chapter provides information for making configuration changes for the following basic features: Wallpaper Screensaver User Password Administrator Password Phone Lock Time and Date Language Softkey Layout Key as Send ...
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The following table lists the wallpaper image format and resolution for each phone model: Wallpaper Image Phone Model Resolution Format SIP-T38G .jpg/.png/.bmp <=480*272 SIP-T32G .jpg/.png/.bmp <=480*272 Procedure The wallpaper shown on the idle screen can be configured using the configuration files or locally.
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Configuring Basic Features Change the wallpaper shown on the idle screen via web user interface. Navigate to: http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Phone-Pref erence.htm Change the wallpaper shown on the idle screen via phone user interface. To upload a customized wallpaper via web user interface: Click on Phone->Preference.
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A screen saver is an animated image that is activated on the IP phone display after periods of user inactivity. The following table lists the screensaver image format and resolution for each phone model: Screensaver Image Phone Model Resolution Format SIP-T38G .jpg/.png/.bmp <=480*272 SIP-T32G .jpg/.png/.bmp <=480*272...
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Configuring Basic Features Procedure The screensaver can be configured using the configuration files or locally. Specify the access URL of the customized screensaver. Configuration File <y0000000000xx>.cfg For more information, refer to Access URL of Screensaver Image on page 312. Upload the customized screensaver.
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Administrator’s Guide for SIP-T3xG IP Phones Click Upload to upload the file. Click Confirm to accept the change. The customized screensaver appears in the pull-down list of Screensaver. To configure the screensaver via web user interface: Click on Phone->Preference.
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Configuring Basic Features Select the desired time from the pull-down list of Screensaver Time. Click Confirm to accept the change. To configure the screensaver via phone user interface: Press Menu->Display->Screensaver. Press , or the Switch soft key to select the desired time in the Time-out field.
Administrator’s Guide for SIP-T3xG IP Phones You can configure the backlight time as one of the following types: Always On: Backlight is turned on permanently. 1min, 2min, 5min, 10min or 30min: Backlight is turned off when the IP phone is ...
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Configuring Basic Features Select the desired value from the pull-down list of Backlight Time. Click Confirm to accept the change. To configure the backlight via phone user interface: Press Menu->Display->Backlight. Press , or the Switch soft key to select the desired level from the Active Level field.
Administrator’s Guide for SIP-T3xG IP Phones Procedure User password can be changed using the configuration files or locally. Change the user password of the IP phone. Configuration File <y0000000000xx>.cfg For more information, refer to on User Password page 236. Change the user password of the IP phone.
Configuring Basic Features if having administrator privileges. The administrator password can be only changed by the administrator. The IP phones support ASCII characters 32-126(0x20-0x7E) only in passwords. Procedure Administrator password can be changed using the configuration files or locally. Change the administrator password of the IP phone.
Administrator’s Guide for SIP-T3xG IP Phones To change the administrator password via phone user interface: Press Menu->Setting->Advanced Settings (password: admin) ->Set Password. Enter the old password in the Current PWD field. Enter the new password in the New PWD field. Enter the new password again in the Confirm PWD field.
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Configuring Basic Features http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Phone-F eatures.htm Assign a keypad lock key. Navigate to: http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Dsskey. Configure the type of phone lock. Phone User Interface Assign a keypad lock key. To configure phone lock via web user interface: Click on Phone->Features->Phone Lock. Select the desired type from the pull-down list of Keypad Lock.
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Administrator’s Guide for SIP-T3xG 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 the type of phone lock via phone user interface: Press Menu->Setting->Advanced Settings (password: admin) ->Phone Setting->Lock.
Configuring Basic Features Time Zone A time zone is a region on the 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 the IP phone to obtain the time and date from the NTP server, you need to set the time zone.
Administrator’s Guide for SIP-T3xG IP Phones Procedure Configuration changes can be performed using the configuration files or locally. Configure the NTP server, time zone and DST. Configure the time and date Configuration File <y0000000000xx>.cfg formats. For more information, refer to Time and Date on page 239.
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Configuring Basic Features Enter the start date in the Start Date field. Enter the start time in the Start Hour of Day field. Select the end month from the pull-down list of Stop Month. Enter the end date in the Stop Date field. Enter the end time in the Stop Hour of Day field.
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Administrator’s Guide for SIP-T3xG IP Phones Select the end week from the pull-down list of Stop Week of Month. Enter the desired offset in the Offset (Minutes) field. Click Confirm to accept the change. To configure the time and date manually via web user interface: Click on Phone->Preference.
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Configuring Basic Features Enter the date and time in the corresponding fields. Click Confirm to accept the change. To configure the time and data format via web user interface: Click on Phone->Preference. Select the desired value from the pull-down list of Time Format.
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Administrator’s Guide for SIP-T3xG IP Phones Select the desired value from the pull-down list of Date Format. Click Confirm to accept the change. To configure the NTP server and time zone via phone user interface: Press Menu->Setting->Basic Settings->Time & Date->SNTP Settings. Press , or the Switch soft key to select the time zone that applies to your area from the Time Zone field.
Configuring Basic Features Press the Save soft key to accept the change. The IP phones support multiple languages. The languages used on the phone user interface and web user interface can be specified respectively as required. The following table lists the languages supported by the phone user interface and the web user interface.
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Administrator’s Guide for SIP-T3xG IP Phones Available Associated Language Pack Language Polish lang-Polish.txt Spanish lang-Spanish.txt Turkish lang-Turkish.txt Procedure Loading language pack can be only performed using the configuration files. Specify the access URL of the language pack. Configuration File <y0000000000xx>.cfg For more information, refer to Language on page 244.
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Configuring Basic Features Select the desired language from the pull-down list of Web Language. Click Confirm to accept the change. To specify the language for the phone user interface via phone user interface: Press Menu->Settings->Basic Settings->Language. Press to select the desired language. Press the Save soft key to accept the change.
Administrator’s Guide for SIP-T3xG IP Phones Softkey layout is used to customize the soft keys at the bottom of the IP phone to best suit the needs of users. It can be controlled based on the call states. In addition to specifying which soft keys to display, you can determine the display order of the soft keys.
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Configuring Basic Features Call State Default Soft Key Optional Soft Key Empty Cancel Transfer Empty Empty Switch SemiAttendTransBack Empty Cancel Transfer Empty Hold Mute Conference SWAP Talk Cancel NewCall Switch Answer Reject Transfer Empty Resume Switch Hold Empty Answer Cancel Reject NewCall Empty...
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Administrator’s Guide for SIP-T3xG IP Phones Call State Default Soft Key Optional Soft Key Cancel Empty Empty Empty Switch InConferenceTalk Conference Cancel Empty Empty Hold Switch Conferenced Split Answer Cancel Reject Mute Procedure Softkey layout can be configured using the configuration files or locally. Specify the access URL of the softkey layout template.
Configuring Basic Features Click to move up or down the soft key. Click Confirm to accept the change. The key as send feature allows assigning the pound key or star key as a send key. The send tone feature determines whether the IP phone plays a key tone when a user presses the send key.
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Administrator’s Guide for SIP-T3xG IP Phones Select the desired value from the pull-down list of Key As Send. Click Confirm to accept the change. To configure the send tone via web user interface: Click on Phone->Features->Audio Settings. Select the desired value from the pull-down list of Send Tone. Click Confirm to accept the change.
Configuring Basic Features To configure the send key via phone user interface: Press Menu->Features->Key as Send. Press , or the Switch soft key to select "#" or "*" from the Key as Send field, or select Disable to disable this feature. Press the Save soft key to accept the change.
Administrator’s Guide for SIP-T3xG IP Phones Enter the hotline number in the Hotline field. Enter the delay time in the Hotline Time-out (seconds) (0~180s) field. Click Confirm to accept the change. To configure hotline via phone user interface: Press Menu->Features->Hot Line. Enter the hotline number in the Hotline field.
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Configuring Basic Features Configure the call log. Navigate to: Web User Interface http://<phoneIPAddress>/cgi-bin/ Local cgiServer.exx?page=Phone-Feat ures.htm Phone User Interface Configure the call log. To configure the call log via web user interface: Click on Phone->Features->General Information. Select the desired value from the pull-down list of Save Call Log. Click Confirm to accept the change.
Administrator’s Guide for SIP-T3xG IP Phones When the IP phone misses calls, the missed call log feature allows the IP phone to display the number of the missed calls and indicator icon on the idle screen, and to log the missed calls in the Missed Calls list. The missed call log feature is configurable on a per-account basis.
Configuring Basic Features 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 directory can be used to store the frequently used contacts. When adding a contact to the local directory, you can specify the account, ring tone and group for the contact in addition to name and phone numbers.
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Administrator’s Guide for SIP-T3xG IP Phones cgiServer.exx?page=Contacts.ht Add the contact to the local Phone User Interface directory directly. To add the contact to the local directory via web user interface: Click on Contacts->Contacts->Contacts. Enter the name and the office, mobile or other numbers in the corresponding fields. Select the desired account from the pull-down list of Account.
Configuring Basic Features Press , or the Switch soft key to select the desired ring tone from the Ring field. Press the Save soft key to accept the change. Commonly, a user dials a number while the IP phone is on-hook, he needs to lift the handset or press the speakerphone key to initiate the call.
Administrator’s Guide for SIP-T3xG IP Phones (If enabled) Enter the desired delay time in the Inter Digit Time (1~14) (seconds) field. Click Confirm to accept the change. The Call waiting feature enables the IP phone to receive a new call when there is an active call.
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Configuring Basic Features cgiServer.exx?page=Phone-Feat ures.htm Configure the call waiting Phone User Interface feature. To configure call waiting via web user interface: Click on Phone->Features->General Information. Select the desired value from the pull-down list of Call Waiting. Click Confirm to accept the change. To configure the call waiting tone via web user interface: Click on Phone->Features->Audio Settings.
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Administrator’s Guide for SIP-T3xG IP Phones Select the desired value from the pull-down list of Call Waiting Tone. Click Confirm to accept the change. To configure call waiting and call waiting tone via phone user interface: Press Menu->Features->Call Waiting. Press , or the Switch soft key to select the desired value from the Call Waiting field.
Configuring Basic Features Procedure Auto redial can be configured using the configuration files or locally. Configure the auto redial feature. Configuration File <y0000000000xx>.cfg For more information, refer to Auto Redial on page 251. Configure the auto redial feature. Navigate to: Web User Interface http://<phoneIPAddress>/cgi-bin/ Local...
Administrator’s Guide for SIP-T3xG IP Phones To configure auto redial via phone user interface: Press Menu->Features->Auto Redial. Press , or the Switch soft key to select the desired value from the Auto Redial field. Enter the desired time in the Auto Redial Interval field. Enter the desired times in the Auto redial times field.
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Configuring Basic Features Select the desired value from the pull-down list of Auto Answer. Click Confirm to accept the change. To configure auto answer via phone user interface: Press Menu->Settings->Advanced Settings (password: admin) ->Accounts. Select the desired account and then press the Enter soft key. Press , or the Switch soft key to select the desired value from the Auto Answer field.
Administrator’s Guide for SIP-T3xG IP Phones Procedure Call completion can be configured using the configuration files or locally. Configure the call completion feature. Configuration File <y0000000000xx>.cfg For more information, refer to Call Completion on page 252. Configure the call completion feature.
CSeq: 1 INVITE Contact: <sip:1012@10.2.8.183:5063> Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T38G 38.70.0.100 Privacy: id Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync P-Preferred-Identity: <sip:1012@10.2.1.199> Content-Length: 302 The anonymous call on code or anonymous call off code configured on the IP phone is used to inform the server of activating or deactivating the anonymous call feature.
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Administrator’s Guide for SIP-T3xG IP Phones m&acc=<x> For T38G, x ranges from 0 to 5. For T32G, x ranges from 0 to 2. Configure the anonymous call Phone User Interface feature. To configure the anonymous call via web user interface: Click on Account->Basic.
Configuring Basic Features Press the Save soft key to accept the change. The anonymous call rejection feature allows the IP phone to automatically reject incoming calls from callers who deliberately block their identities from showing up. The anonymous caller’s phone LCD screen presents ―Anonymity Disallowed‖. The anonymous call rejection on code or anonymous call rejection off code configured on the IP phone is used to inform the server of activating or deactivating the anonymous call rejection feature.
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Administrator’s Guide for SIP-T3xG IP Phones (Optional.) Enter the anonymous call rejection off code in the Off Code field. Click Confirm to accept the change. To configure anonymous call rejection via phone user interface: Press Menu->Features->Anonymous Call. Press , or the Switch soft key to select the desired line from the Line ID field.
Configuring Basic Features Do Not Disturb (DND) allows the IP phone to ignore incoming calls. A user can activate or deactivate the DND feature using a DND soft key or DND key. DND activated on the IP phone disables the local call forward settings. The DND configurations on the IP phone may be overridden by the server settings.
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Administrator’s Guide for SIP-T3xG IP Phones Navigate to: http://<phoneIPAddress>/cgi-bin/c giServer.exx?page=Phone-Featur es.htm Phone User Interface Assign a DND key. To configure a DND key via web user interface: Click on DSS Key->Memory Key (or Line Key). In the desired DSS key field, select DND from the pull-down list of Type. Click Confirm to accept the change.
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Configuring Basic Features Enter the DND off code in the DND Off Code field. Click Confirm to accept the change. To specify the DND authorized numbers via web user interface: Click on Phone->Features> Do Not Disturb. Enter the numbers in the Authorized Numbers (comma separated) field. Multiple numbers are separated by comma.
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Administrator’s Guide for SIP-T3xG IP Phones To specify the return code and the reason via web user interface: Click on Phone->Features->General Information. Select the desired value 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->Features->DSS Keys->Memory Keys (or Line Keys).
Configuring Basic Features For more information, refer to Busy Tone Delay on page 257. Configure the busy tone delay feature. Navigate to: Local Web User Interface http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Phone-Feat ures.htm To configure busy tone delay via web user interface: 1. Click on Phone->Features->General Information. 2....
Administrator’s Guide for SIP-T3xG IP Phones Procedure Return code when refusing a call can be configured using the configuration files or locally. Configure the return code when refusing a call. Configuration File <y0000000000xx>.cfg For more information, refer to Return Code When Refuse page 257.
Configuring Basic Features 180 ring workaround defines whether to deal with the 180 message received after the 183 message. When the caller receives a 183 message, it suppresses any local ringback tone and begins to play the media received. 180 ring workaround allows the IP phone to resume and play the local ringback tone upon a subsequent 180 message received.
Administrator’s Guide for SIP-T3xG 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 the SIP request messages from the IP phone will be forced to send to the outbound proxy server.
Configuring Basic Features ures.htm To specify whether to use outbound proxy server in a dialog via web user interface: Click on Phone->Features->General Information. Select the desired value from the pull-down list of Use Outbound Proxy in Dialog. Click Confirm to accept the change. The IP phones support to configure SIP session timers T1, T2 and T4.
Administrator’s Guide for SIP-T3xG IP Phones Session Timer on page 259. Configure the SIP session timer feature. Navigate to: http://<phoneIPAddress>/cgi-bin/ Local Web User Interface cgiServer.exx?page=Account-Ad v.htm&acc=<x> For T38G, x ranges from 0 to 5. For T32G, x ranges from 0 to 2. To configure the session timer via web user interface: Click on Account->Basic.
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Configuring Basic Features The IP phones support to use session timer to send periodic re-INVITE requests to refresh the session during a call. The session timer is defined in RFC 4082. The IP phones support two refresher modes: UAC and UAS. The UAC mode means refreshing the session from the client, while the UAS mode means refreshing the session from the server.
Administrator’s Guide for SIP-T3xG IP Phones Select the desired refresher from the pull-down list of Refresher. Click Confirm to accept the change. Call hold feature 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. The IP phones support two call hold methods, one is RFC 3264, it is used to set the ―a‖...
Configuring Basic Features Navigate to: http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Phone-Feat ures.htm To configure the call hold method via web user interface: Click on Phone->Features->General Information. Select the desired value from the pull-down list of RFC 2543 Hold. Click Confirm to accept the change. Call forward allows redirecting an incoming call to a third party.
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Administrator’s Guide for SIP-T3xG IP Phones Procedure Call forward can be configured locally. Configure the call forward feature. Navigate to: Web User Interface http://<phoneIPAddress>/cgi- Local bin/cgiServer.exx?page=Pho ne-Features.htm Configure the call forward Phone User Interface feature. To configure always forward via web user interface: Click on Phone->Features->Forward.
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Configuring Basic Features (Optional.) Enter the on code or off code in On Code or Off Code field. Click Confirm to accept the change. To configure no answer forward via web user interface: Click on Phone->Features->Forward. Mark the desired radio box in the No Answer field. Select the ring time to wait before forwarding from the pull-down list of After Ring Time (seconds).
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Administrator’s Guide for SIP-T3xG IP Phones (Optional.) Enter the on code or off code in On Code or Off Code field. Click Confirm to accept the change. To configure call forward via phone user interface: Press Menu->Features->Call Forward. Press to select the desired forwarding type, and then press the Enter soft key.
Configuring Basic Features 2) Enter the destination number you want to forward all unanswered incoming calls to in the Forward to field. 3) Press , or the Switch soft key to select the ring time to wait before forwarding from the After Ring Times field. The default ring time is 60 seconds.
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Administrator’s Guide for SIP-T3xG IP Phones Configure the semi-attended transfer feature. Navigate to: http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Phone-Feat ures.htm To configure call transfer via web user interface: Click on Phone->Features->Transfer Settings. Select the desired values from the pull-down lists of Semi-Attended Transfer, Blind Transfer on Hook and Attended Transfer on Hook.
Configuring Basic Features For more information, refer to Network Conference on page 263. Configure the network conference. Navigate to: http://<phoneIPAddress>/cgi-bin/ Local Web User Interface cgiServer.exx?page=Account-Ad v.htm&acc=<x> For T38G, x ranges from 0 to 5. For T32G, x ranges from 0 to 2. To configure the network conference via web user interface: Click on Account->Basic.
Administrator’s Guide for SIP-T3xG IP Phones For local conference, all parties release the call when the conference initiator drops the conference call. The transfer on conference hang up feature allows the other two parties remain connected when the conference initiator drops the conference call. Procedure Transfer on conference hang up feature can be configured using the configuration files or locally.
Configuring Basic Features Direct pickup is used for picking up an incoming call on a specific extension. A user can pick up the incoming call using a direct pickup key or DPickup soft key. This feature depends on support from a SIP server. For many SIP servers, direct pickup is implemented requiring a direct pickup code.
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Administrator’s Guide for SIP-T3xG IP Phones http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Account -Adv.htm&acc=<x> For T38G, x ranges from 0 to 5. For T32G, x ranges from 0 to 2. Phone User Interface Assign a direct pickup key. To configure a direct pickup key via web user interface: Click on DSS Key->Memory Key (or Line Key).
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Configuring Basic Features Enter the direct call pick up code in the Direct Call Pickup Code field. Click Confirm to accept the change. To configure the direct call pickup code on a per-account basis via web user interface: Click on Account->Basic. Select the desired account from the pull-down list of Account.
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Administrator’s Guide for SIP-T3xG IP Phones Enter the direct pickup feature code in the Direct Call Pickup Code field. Click Confirm to accept the change. To configure the direct pickup key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key.
Configuring Basic Features Procedure Group pickup can be configured using the configuration files or locally. Configure the group pickup code on a per-account basis. <MAC>.cfg For more information, refer to Group Pickup on page 266. Assign a group pickup key. Configuration File For more information, refer to Group Pickup Key...
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Administrator’s Guide for SIP-T3xG IP Phones Select the desired line from the pull-down list of Account. Click Confirm to accept the change. To configure the group call pickup feature on a phone basis via web user interface: Click on Phone->Features->Call Pickup. Select the desired value from the pull-down list of Group Call Pickup.
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Configuring Basic Features Click on Advanced. Enter the group call pickup code in the Group Call Pickup Code field. Click Confirm to accept the change. To configure a group pickup key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key.
Configuring Basic Features Select the desired value from the pull-list of Dialog-Info Call Pickup. Click Confirm to accept the change. Call return, also known as last call return, provides convenience for a user to place a call back to the caller of the last incoming call. The IP phones implement call return using a call return key.
Administrator’s Guide for SIP-T3xG IP Phones 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->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key.
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Configuring Basic Features http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Dsskey. Phone User Interface Assign a call park key. To configure a call park key via web user interface: Click on DSS Key->Memory Key (or Line Key). In the desired DSS key field, select Call Park from the pull-down list of Type. Enter the desired value (e.g., call park feature code) in the Value field.
Administrator’s Guide for SIP-T3xG IP Phones interface. The IP phones support both HTTP and HTTPS protocols for accessing the web user interface. HTTP is an application protocol that runs on top of the TCP/IP suite of protocols. HTTPS is a web protocol that encrypts and decrypts user page requests as well as the pages returned by the web server.
Configuring Basic Features Select the desired type from the pull-down list of Type. Click Confirm to accept the change. The web user interface pops up a dialog box 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->Settings->Advanced Settings (password: admin) ->Network->Webserver Type.
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Administrator’s Guide for SIP-T3xG IP Phones should be preferentially displayed. Procedure CLIP can be configured using the configuration files or locally. Configure the presentation of the caller identity. Configuration File <MAC>.cfg For more information, refer to Calling Line Identification Presentation on page 269.
Configuring Basic Features Click Confirm to accept the change. Connected Line Identification Presentation (COLP) allows the IP phone to display the identity of the callee specified for outgoing calls. The IP phone can display the Dialed Digits, or the identity in a SIP header (Remote-Party-ID or P-Asserted-Identity) received, or the identity in the From header carried in the UPDATE message sent by the callee as described in RFC 4916.
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Administrator’s Guide for SIP-T3xG IP Phones There are 3 common methods of transmitting DTMF digits on SIP calls: RFC 2833 – DTMF digits are transmitted by RTP Events compliant to RFC 2833. INBAND – DTMF digits are transmitted in the voice band. ...
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Configuring Basic Features DTMF on page 270. Configure the method of transmitting DTMF digits and the payload type. Navigate to: http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Account -Adv.htm&acc=<x> For T38G, x ranges from 0 to 5. Local Web User Interface For T32G, x ranges from 0 to 2. Configure the number of times for the IP phone to send the end RTP Event packet.
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Administrator’s Guide for SIP-T3xG IP Phones Enter the desired value in the DTMF Payload field. Click Confirm to accept the change. To configure the number of times to send the end RTP Event packet via web user interface: Click on Phone->Features->General Information.
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Configuring Basic Features Select the desired value (1-3) from the pull-down list DTMF Repetition. Click Confirm to accept the change. Intercom allows establishing a two-way audio conversation directly. The called phone picks up intercom calls automatically and establishes intercom conversations. This feature depends on support from a SIP server.
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Administrator’s Guide for SIP-T3xG IP Phones http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Dsskey.htm Phone User Interface Assign an intercom key. To configure an intercom key via web user interface: Click on DSS Key->Memory Key (or Line Key). In the desired DSS key field, select Intercom from the pull-down list of Type. Enter the remote extension number in the Value field.
Configuring Basic Features Accept Intercom Accept Intercom allows the IP phone to automatically answer an incoming intercom call. Intercom Mute Intercom Mute allows the IP phone to mute the microphone for incoming intercom calls. Warning Tone Warning Tone allows the IP phone to play a warning tone before answering an intercom call.
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Administrator’s Guide for SIP-T3xG IP Phones Select the desired values from the pull-down lists of Accept Intercom, Intercom Mute, Warning Tone and Intercom Barge. Click Confirm to accept the change. To configure intercom via phone user interface: Press Menu->Features->Intercom. Press , or the Switch soft key to select the desired values from the Accept Intercom, Intercom Mute, Warning Tone and Intercom Barge fields.
Configuring Advanced Features This chapter provides information for making configuration changes for the following advanced features: Distinctive Ring Tones Tones Remote Phonebook LDAP Busy Lamp Field BLF List Shared Call Appearance As-Feature-Event Automatic Call Distribution ...
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Administrator’s Guide for SIP-T3xG IP Phones Alert-Info: http://127.0.0.1/Bellcore-dr 192.168.0.12:8080/ring.wav Family Alert-Info: <http:// >;info= ;x-line-id=0 If the ―Alter-Info‖ header contains the keywords ―Bellcore-drN‖ or ―MyMelodyN‖, the IP phone will map the index ―N‖ to the relevant ring tone. Value of N Ring Tone Ring1.wav Ring2.wav...
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Configuring Advanced Features Navigate to: http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Phone-R ing.htm To configure the internal ringer text and internal ringer file via web user interface: Click on Phone->Ring. Enter the keywords in the Internal Ringer Text fields. Select the desired ring tones for each text from the pull-down lists of Internal Ringer File.
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Administrator’s Guide for SIP-T3xG IP Phones Czech Denmark Finland France Germany Great Britain Greece Hungary Lithuania India Italy Japan Mexico New Zealand Netherlands Norway Portugal ...
Configuring Advanced Features Condition Description Stutter When receiving a voice mail Message When receiving a text message Auto Answer When automatically answering a call Procedure Tones can be configured using the configuration files or locally. Configure the tones for the IP phone.
Administrator’s Guide for SIP-T3xG IP Phones Remote phonebook is the IP phone book maintained centrally, which is stored on the remote server. Users just need the access URL of the remote phonebook. The IP phone can establish a connection with the remote server and download the entries, and then display the entries on the phone user interface.
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Configuring Advanced Features Specify how often the IP phones refresh the local cache of the remote phonebook. Navigate to: http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Phone-F eatures.htm To specify the access URL of the remote phonebook via web user interface: Click on Contacts->Remote Phone Book. Enter the access URL in the Phone Book URL field.
Administrator’s Guide for SIP-T3xG IP Phones Enter the desired time in the Load Remote Phone Book Interval (120~2592000s) field. Click Confirm to accept the change. LDAP (Lightweight Directory Access Protocol) is an application protocol for accessing and maintaining information services of the distributed directory over an IP network. The IP phone can be configured to interface with a corporate directory server that supports LDAP version 2 or 3 (Microsoft’s Active Directory is included).
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Configuring Advanced Features be shown according to your query setup. Assign a DSS key to be an LDAP key, and press the LDAP key to enter the LDAP Search interface when the IP phone is idle. LDAP Attributes The following table lists the most common attributes used to configure the LDAP lookup on the IP phone: Abbreviation Name...
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Administrator’s Guide for SIP-T3xG IP Phones n/cgiServer.exx?page=Contact s-LDAP .htm Phone User Interface Assign an LDAP key. To configure LDAP via web user interface: Click on Contacts->LDAP. Enter the values in the corresponding fields. Select the desired values from the corresponding pull-down lists. Click Confirm to accept the change.
Configuring Advanced Features In the desired DSS key field, select LDAP from the pull-down list of Type. Click Confirm to accept the change. To configure an LDAP key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field.
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Administrator’s Guide for SIP-T3xG IP Phones Procedure BLF can be configured using the configuration files or locally. Assign a BLF key. For more information, refer to BLF Key on page 328. Configuration File y0000000000xx.cfg Specify whether to use visual and audio alert for BLF pickup. For more information, refer to on page 283.
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Configuring Advanced Features (Optional.) Enter the pickup code in the Extension field. Click Confirm to accept the change. To configure the visual and audio alert feature via web user interface: Click on Phone->Features->Call Pickup. Select the desired value from the pull-down list of Visual Alert for BLF Pickup. Select the desired value from the pull-down list of Audio Alert for BLF Pickup.
Administrator’s Guide for SIP-T3xG IP Phones Press , or the Switch soft key to select BLF from the Type field. Press , or the Switch soft key to select the desired line from the Account ID field. Enter the phone number or extension you want to monitor in the Value field. (Optional.) Enter the pickup code in the Extension field.
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Configuring Advanced Features n/cgiServer.exx?page=Dsskey. Phone User Interface Assign BLF list keys. To configure BLF list via web user interface: Click on Account->Basic. Select the desired account from the Account field. Click on Advanced. Enter the BLF List URI in the BLF List URI field. (Optional.) Enter the BLF pickup code in the BLF List Pickup Code field.
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Administrator’s Guide for SIP-T3xG IP Phones Repeat steps 2 to 3 to configure more BLF List keys. Click Confirm to accept the change. After the above configurations, according to the response message from the BLF List server, the IP phone will automatically assign the phone number of the BLF List users to the BLF List keys in order.
Configuring Advanced Features not all of them. An SCA user can retrieve a public hold call on the shared line. If the SCA bridging feature is enabled, SCA users can barge in an existing call on the shared line. Procedure Register the primary and secondary lines on two IP phones using the configuration files or locally.
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Administrator’s Guide for SIP-T3xG IP Phones Select BroadSoft SCA from the pull-down list of Shared Line. Click Confirm to accept the change. To assign a shared line key via web user interface: Click on DSS Key->Memory Key (or Line Key). In the desired DSS key field, select Shared Line from the pull-down list of Type.
Configuring Advanced Features Click Confirm to accept the change. To assign a shared line key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key. Press , or the Switch soft key to select Shared Line from the Type field. Press , or the Switch soft key to select the desired line from the Account ID field.
Administrator’s Guide for SIP-T3xG IP Phones To configure the as-feature-event via web user interface: Click on Phone->Features->General Information. Select the desired value from the pull-down list of Feature Synchronization. Click Confirm to accept the change. Automatic Call Distribution (ACD) enables organizations to manage a large number of phone calls on an individual basis.
Configuring Advanced Features Assign an ACD key. Navigate to: Web User Interface Local http://<phoneIPAddress>/cgi-bin/ cgiServer.exx?page=Dsskey.htm Phone User Interface Assign an ACD key. To configure an ACD key via web user interface: Click on DSS Key->Memory Key (or Line Key). In the desired DSS key field, select ACD from the pull-down list of Type. Select the desired line from the pull-down list of Account.
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Administrator’s Guide for SIP-T3xG IP Phones audio and visual MWI when receiving new voice messages. The IP phones support both solicited and unsolicited MWI. Unsolicited MWI is a server related feature. Solicited MWI: MWI notification is subscription-based. The IP phone sends a SUBSCRIBE message to the server for message-summary updates.
Configuring Advanced Features Enter the period time in the MWI Subscription Period (Scope: 0~84600) (seconds) field. Click Confirm to accept the change. Call recording enables a user to record a call. It depends on support from a SIP server. When the user presses a call record key, the IP phone sends a record request to the server.
The example of an HTTP GET message for reference: 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-T38G 38.70.0.100 00:16:65:11:30:68\r\n If the recording is successfully started, the server will respond with 200 OK as below: <YealinkIPPhoneText> <Title> </Title> <Text>...
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Configuring Advanced Features <YealinkIPPhoneText> If the recording fails for some reasons, for example, the recording box is full, the server will respond with 200 OK as below: <YealinkIPPhoneText> <Title> </Title> <Text> Probably the recording box is full. </Text> <YealinkIPPhoneText> When the user presses the URL record key for the second time, the IP phone sends an HTTP GET message to the server, and then the server will respond with the following 200 OK message: <YealinkIPPhoneText>...
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Administrator’s Guide for SIP-T3xG IP Phones In the desired DSS key field, select Record from the pull-down list of Type. Click Confirm to accept the change. To configure a URL Record key via web user interface: Click on DSS Key->Memory Key (or Line Key). In the desired DSS key field, select URL Record from the pull-down list of Type.
Configuring Advanced Features Press , or the Switch soft key to select Key Event from the Type field. Press , or the Switch soft key to select Record from the Key Type field. Press the Save soft key to accept the change. To configure a URL Record key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys).
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Administrator’s Guide for SIP-T3xG IP Phones Leave the Value field blank. Click Confirm to accept the change. To configure a hot desking key via phone user interface: Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). Select the desired DSS key. Press , or the Switch soft key to select Key Event from the Type field.
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Configuring Advanced Features Event Description Log On When the IP phone successfully registers an account. Log Off When the IP phone logs off the registered account. Register Failed When the IP phone fails to register an account. Off Hook When the IP phone is off hook. On Hook When the IP phone is on hook.
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Administrator’s Guide for SIP-T3xG IP Phones Event Description When the state of the IP phone changes from idle to Idle to Busy busy. Busy to Idle When the state of phone changes from busy to idle. The following table lists the variable values used when specifying a URL: Variable Description $mac...
Configuring Advanced Features Procedure Action URL can be configured using the configuration files or locally. Configure the action URL on the IP phone. Configuration File <y0000000000xx>.cfg For more information, refer to Action URL on page 290. Configure the action URL on the IP phone.
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Administrator’s Guide for SIP-T3xG IP Phones phone will perform the specified action and respond with a 200 OK message. The HTTP GET request may contain variable named as ―key‖ and variable value, which are separated by ―=‖. The URI format is: http://IP address of phone/cgi-bin/ cgiServer.exx?key= variable value (e.g.
Configuring Advanced Features HTTP GET request from the specified IP address, the phone LCD screen prompts the message ―Allow Remote Control?‖. You can specify one or more trusted IP addresses on the IP phone. You can also configure the IP phone to receive and handle the URI from any IP address.
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Administrator’s Guide for SIP-T3xG IP Phones Many SIP servers are deployed in redundant pairs, designated as primary and secondary servers. The IP phone must always contact the primary server except in failover conditions. 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.
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Configuring Advanced Features The IP phone picks the first record, because its order of 90 is lower than 100. The pref parameter is unimportant as there is no other record with order 90. The flag ―s‖ indicates performing the SRV query next. TCP will be used, targeted to a host determined by an SRV query of ―_sip._tcp.example.com‖.
Administrator’s Guide for SIP-T3xG IP Phones The following figure illustrates the IP phone has the availability of performing DNS SRV query, and fails over the request to the secondary server when there is no response from the primary server. Procedure DNS SRV query can be configured using the configuration files or locally.
Configuring Advanced Features Select DNS-SRV from the pull-down list of Transport. Click Confirm to accept the change. LLDP (Linker Layer Discovery Protocol) is a vendor-neutral Link Layer protocol. It allows IP phones to receive and/or transmit device-related information to directly connected devices on the network that are also using the protocol, and store the information that is learned about other devices.
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The default value is 120s. End of LLDPDU Marks end of LLDPDU. Name assigned to the IP phone. System Name The default value is ―yealink‖. Description of the IP phone. System Description The default value is ―yealink‖. The supported and enabled capabilities of phone.
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Inventory – Serial Serial number of phone. Number Inventory – Manufacturer name of phone. Manufacturer Name The default value is ―yealink‖. Inventory – Model Model name of phone. Name Assertion identifier of phone. Asset ID The default value is ―asset‖.
Administrator’s Guide for SIP-T3xG IP Phones The valid values range from 1 to 3600. Click Confirm to accept the change. The web user interface pops up a dialog box to prompt that the settings will take effect after reboot. Click OK to reboot the IP phone. VLAN (Virtual Local Area Network) is used to logically divide a physical network into several broadcast domains.
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Configuring Advanced Features The IP phones support to configure the VLAN information either manually or dynamically using the LLDP feature. For more information on LLDP , refer to LLDP on page 157. Note The VLAN information in the received LLDP packets will override the manual configuration.
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Administrator’s Guide for SIP-T3xG IP Phones Select the desired value (0-7) from the pull-down list of Priority. Click Confirm to accept the change. The web user interface pops up a dialog box to prompt reboot to make the settings effective. Click OK to reboot the IP phone.
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Configuring Advanced Features Select the desired value (0-7) from the pull-down list of Priority. Click Confirm to accept the change. The web user interface pops up a dialog box to prompt that the settings will take effect after reboot. Click OK to reboot the IP phone. To configure VLAN for Internet port (or PC port) via phone user interface: Press Menu->Settings->Advanced Settings (password: admin) ->Network->VLAN->WAN Port (or PC Port).
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Administrator’s Guide for SIP-T3xG IP Phones There are two types of VPN: remote-access VPN (connecting an individual device to a network) and site-to-site VPN (connecting two networks together). Remote-access VPNs allow employees to access their company's intranet from home or outside the office, and site-to-site VPNs allow employees in geographically separated offices to share one cohesive virtual network.
Configuring Advanced Features Click Browse to locate the tar package from the local system. Click Import to import the tar package. Select the desired value from the pull-down list of VPN Active after importing. Click Confirm to accept the change. The web user interface pops up a dialog box to prompt that the settings will take effect after reboot.
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Administrator’s Guide for SIP-T3xG IP Phones consider when configuring a modern QoS implementation, these include: bandwidth, delay, jitter and loss. QoS provides better network service by providing the following features: Supporting dedicated bandwidth Improving loss characteristics Avoiding and managing network congestion ...
Configuring Advanced Features Voice QoS For VoIP transmissions to be intelligible to the receiver, voice packets should not be dropped, excessively delayed, or suffer varying delay. DiffServ model can guarantee high-quality voice transmission when the voice packets are configured higher DSCP value.
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Administrator’s Guide for SIP-T3xG IP Phones Enter the desired value (0-63) in the SIP QoS field. Click Confirm to accept the change. The web user interface pops up a dialog box to prompt that the settings will take effect after reboot. Click OK to reboot the IP phone.
Configuring Advanced Features allows applications to operate behind a NAT to discover the presence of the network address translator, and obtain the mapped (public) IP address and port number that the NAT has allocated for the UDP connections to remote parties. The protocol requires assistance from a third-party network server (STUN server) usually located on public Internet.
Administrator’s Guide for SIP-T3xG IP Phones Enter the IP address or the domain name in the STUN Server field. Click Confirm to accept the change. IEEE 802.1X authentication is an IEEE Standard for Port-based Network Access Control (PNAC). It is part of the IEEE 802.1 group of networking protocols. It provides an authentication mechanism to devices wishing to attach to a LAN or WLAN.
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Configuring Advanced Features Configure the 802.1X authentication on the IP phone. Navigate to: Web User Interface http://<phoneIPAddress>/cgi-bi Local n/cgiServer.exx?page=Networ k-Adv.htm Configure the 802.1X Phone User Interface authentication on the IP phone. To configure the 802.1X via web user interface: Click on Network->Advanced. Select the desired value from the pull-down list of 802.1x Mode.
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Administrator’s Guide for SIP-T3xG IP Phones Press , or the Switch soft key to select the desired value from the 802.1x Mode field. (If EAP-MD5 is selected) Enter the username for authentication in the Identity field. (If EAP-MD5 is selected) Enter the password for authentication in the MD5 Password field.
Configuring Audio Features This chapter provides information for making configuration changes for the following audio features: Audio Codecs Acoustic Clarity Technology CODEC is an abbreviation of COmpress-DECompress. It is 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.
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Administrator’s Guide for SIP-T3xG IP Phones Codec Configuration Method Priority RTPmap Configuration Files G722 Web User Interface Configuration Files G723_53 Web User Interface Configuration Files G723_63 Web User Interface Configuration Files G726_16 Web User Interface Configuration Files G726_24 Web User Interface Configuration Files G726_32 Web User Interface...
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Configuring Audio Features Audio Codecs on page 299. Configure the codecs and adjust the priority of the enabled codecs on a per-account basis. Navigate to: http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Account -Codec.htm&acc=<x> For T38G, x ranges from 0 to 5. Local Web User Interface For T32G, x ranges from 0 to 2.
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Administrator’s Guide for SIP-T3xG IP Phones Click to adjust the priorities of the enabled codecs. Click Confirm to accept the change. To configure the Ptime on a per-account basis via web user interface: Click on Account->Basic. Select the desired account from the pull-down list of Account. Click on Advanced.
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.
Administrator’s Guide for SIP-T3xG 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 can also be used to deactivate some processes during non-speech section of an audio session.
Configuring Audio Features Comfort Noise Generation (CNG) is used to generate background noise for voice communications during periods of silence that occur during the conversation. It is part of the silence suppression or VAD handling for VoIP technology. CNG, in conjunction with VAD algorithms, quickly determines when periods of silence occur and inserts artificial noise until voice activity resumes.
Administrator’s Guide for SIP-T3xG IP Phones Jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in evenly spaced intervals. Jitter is variations in packet arrival time, can occur because of network congestion, timing drift or route changes. The jitter buffer, which is located at the receiving end of the voice connection, intentionally delays the arriving packets so that the end user experiences a clear connection with very little sound distortion.
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Configuring Audio Features Enter the fixed delay time for fixed jitter buffer in the Normal Delay field. Click Confirm to accept the change.
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Administrator’s Guide for SIP-T3xG IP Phones...
Configuring Security Features This chapter provides information for making configuration changes for the following security-related features: Transport Layer Security Secure Real-Time Transport Protocol Encrypting Configuration Files The TLS protocol is a commonly-used protocol for providing communications privacy and managing the security of message transmission.
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Administrator’s Guide for SIP-T3xG IP Phones The IP phone can encrypt SIP with TLS, which is called SIPS. When TLS is enabled for an account, the SIP message of this account will be encrypted, and a lock icon appears on the phone LCD screen after the successful TLS negotiation.
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Configuring Security Features Upload certificates to the IP phone. For more information, refer to <y0000000000xx>.cfg Uploading Certificates on page 306. Configure the IP phone to use TLS. Navigate to: http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Account .htm&acc=<x> For T38G, x ranges from 0 to 5. For T32G, x ranges from 0 to 2.
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Administrator’s Guide for SIP-T3xG IP Phones Select TLS from the pull-down list of the Transport. Click Confirm to accept the change. To Specify the IP phone to encrypt the SIP signal using RC4 encryption algorithm: Click on Account->Basic. Select the desired account from the pull-down list of Account. Click on Advanced.
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Configuring Security Features Enter the desired key in the Signal Encode Key field. Click Confirm to accept the change. To configure Only Accepted Trusted Certificates via web user interface: Click on Security->Trusted Certificates.
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Administrator’s Guide for SIP-T3xG IP Phones Select the desired value from the pull-down list of Only Accept Trusted Certificates. Click Confirm to accept the change. To upload the trusted certificate via web user interface: Click on Security->Trusted Certificates. Click Browse to select the trusted certificate (*.crt or *.cer) from your local system. Click Upload to upload the trusted certificate.
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Configuring Security Features The web user interface pops up the dialog box to prompt ―Rebooting, please wait…‖. Secure Real-Time Transport Protocol (SRTP) provides means of encrypting the RTP streams during VoIP phone calls to avoid interception and eavesdropping. The parties participating in the call should enable the SRTP feature simultaneously.
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Administrator’s Guide for SIP-T3xG IP Phones The sample of the RTP encryption algorithm carried in the SDP of the 200 OK message for reference: m=audio 11780 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NGY4OGViMDYzZjQzYTNiOTNkOWRiYzRlMjM0Yzcz a=sendrecv a=ptime:20 a=fmtp:101 0-15 You can configure the SRTP feature on a per-account basis.
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Note Yealink also supplies an encryption tool (EncryptUtilityLinux.exe) to support Linux platforms if required. You can also encrypt the configuration files using the Yealink Configuration Conversion Tool. For more information, refer to the document ―Yealink Configuration Conversion Tool User Guide‖.
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Administrator’s Guide for SIP-T3xG IP Phones decrypting before provisioning. Procedure to Encrypt Configuration Files To encrypt the <y0000000000xx>.cfg file: Place the ―EncryptUtilityWindows.exe‖ tool and <y0000000000xx>.cfg file to the same directory (i.e., D:\). Open a command line window application (i.e., DOS window). Enter the following command, and then press the <Enter>...
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Configuring Security Features Enter the values in the Common AES Key and MAC-Oriented AES Key fields. Click Confirm to accept the change.
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Administrator’s Guide for SIP-T3xG IP Phones...
38.x.x.x.rom SIP-T32G 32.x.x.x.rom Note You can download the latest firmware at: http://www.yealink.com/Support.aspx. Upgrade via Web User Interface To manually upgrade firmware via web user interface, you need to store the firmware to your local system in advance. To upgrade the firmware manually via web user interface: Click on Phone->Upgrade.
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Administrator’s Guide for SIP-T3xG IP Phones The web user interface pops up the dialog box to prompt ―Firmware of the SIP PHONE will be updated. It will take several minutes to complete. So, please don't power off!‖. Click OK to confirm the upgrading. Note Do not unplug the network and power cables when the IP phone is upgrading the firmware.
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Upgrading the Firmware configuration files. Specify the access URL of the firmware. For more information, refer to Upgrading the Firmware page 308. Configure the way for the IP phone to check for configuration files. Local Web User Interface Navigate to: http://<phoneIPAddress>/cgi-bi n/cgiServer.exx?page=Phone- AutoProvision.htm...
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Administrator’s Guide for SIP-T3xG IP Phones (If the Weekly On radio box is marked) Check the desired checkbox in the Day of week field. Click Confirm to accept the change. When the ―Check New Config‖ is set to On, the IP phone will check for both firmware and configuration files stored on the provisioning server during booting up.
Resource Files When configuring some features, you may need to upload resource files to the IP phone. The resources files can be local contact directory, remote phonebook and so on. If the resource file is to be used for all IP phones of the same model, the access URL of the resource file is best specified in the <y0000000000xx>.cfg file.
Administrator’s Guide for SIP-T3xG IP Phones Procedure Use the following procedures to customize a replace rule template. Customizing 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=""...
Resource Files Procedure Use the following procedures to customize dial-now template. Customizing 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=""...
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Administrator’s Guide for SIP-T3xG IP Phones <Default> indicates the start of the default soft key list and </Default> indicates the end of the default soft key list, the default soft keys are displayed on the phone LCD screen by default. Procedure Use the following procedures to customize a soft key layout template.
Resource Files You can add contact one by one on the IP phone directly. In some cases, you may want to add multiple contacts to the IP phone at the same time or share the contacts on many IP phones. You can create a local contact file, and then place the local contact file to the root directory of the provisioning server, specify the access URL of the contact file in the configuration files.
Administrator’s Guide for SIP-T3xG IP Phones starting on a separate line: <group name=‖‖ ring=‖‖/> Where: group name=‖‖ specifies the name of the group. ring=‖‖ specifies the desired ring tone for this group. Specify the values within double quotes. Place this file to the root directory of the provisioning server. The following is an example of a local contact file: <contactData>...
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Resource Files Procedure Use the following procedures to customize an XML phonebook. Customizing an XML phonebook: Open the template file using an ASCII editor. For each contact that you want to add, add the following strings to the IP phonebook, each starting on a separate line: Mary <Name>...
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Administrator’s Guide for SIP-T3xG IP Phones Template on page 311. Configure the access URL of the dial-now rule template. Configuration File <y0000000000xx>.cfg For more information, refer to Access URL of Dial-now Template on page 312. Configure the access URL of the softkey layout template.
Troubleshooting This chapter provides an administrator with general information for troubleshooting some most common problems that may encounter while using the SIP-T3xG IP phones. The IP phone can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which helps an administrator quickly find out the cause of failure and do the troubleshooting more easily.
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Administrator’s Guide for SIP-T3xG IP Phones Select the desired level from the pull-down list of Log Level. Click Confirm to accept the change. The web user interface pops up a dialog box to prompt ―Do you want to restart your machine?‖...
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Troubleshooting Mark the Local radio box in the Export System Log field. Click Export to open file download window, and then save the file to your local system. You can capture packets in two ways: capturing the packets via web user interface or using the Ethernet software.
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Administrator’s Guide for SIP-T3xG IP Phones Click Export to open file download window, and then save the file to your local system. To capture packets using the Ethernet software: Connect the IP phone’s Internet port with the PC to the same HUB, and then use Sniffer, Ethereal or Wireshark software to capture the packets.
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Troubleshooting Select the desired value from the pull-down list of Watch Dog. Click Confirm to accept the change. In some cases, the status indicators consist of power LED, message key indicator, line key indicator, headset key indicator and the on-screen icon/error messages, which are useful for you to figure out the cause of your phone’s failure.
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This section describes solutions to some most common problems that may occur while using the IP phone. If you encounter a problem which is not listed in this section, contact your Yealink reseller for further support. Do one of the followings: Check that the power LED is on to ensure the IP phone is powered on.
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Troubleshooting Ensure that the switch or hub in your network is operational. ― ‖ The phone LCD screen prompts ―No Service‖ message when there is no any available SIP account on the IP phone. Do one of the followings: Confirm if any account is actively registered on the IP phone at the path ...
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Administrator’s Guide for SIP-T3xG IP Phones sensitive microphones and cause feedback. Intermittent voice is mainly caused by packet loss and jitter. Packet loss may due to network congestion. Jitter is mainly due to message recombination of transmission or receiving equipment, such as timeout handling, retransmission mechanism or buffer under run.
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DOB file. There is a tool for you to convert a BMP file to DOB. For more information, refer to the document ―Yealink Auto Provisioning User Guide‖. A factory reset can restore the original password. Please try to long press the OK key when the IP phone is idle, which should lead you to make a factory reset.
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Administrator’s Guide for SIP-T3xG IP Phones the IP phone is PnP enabled, it broadcast the PNP subscribe message to obtain a provisioning server address during booting up, any SIP server recognizing the message will respond with the preconfigured provisioning server address, so the IP phone will be able to download the CFG files from that server address.
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Troubleshooting Do one of the followings: Try to set another available IP address for the IP phone. Check the configuration of the network via phone user interface at the path Menu->Settings->Advanced Settings->Network->WAN Port. If Static IP Client is selected, select DHCP IP Client instead.
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Administrator’s Guide for SIP-T3xG IP Phones The web user interface prompts the message ―Do you want to reset to factory?‖. Click OK to confirm the resetting.
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 provides an authentication mechanism to devices wishing to attach to a LAN or WLAN. ACD (Automatic Call Distribution)--used to distribute calls from large volumes of incoming calls to the registered IP phone users.
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Administrator’s Guide for SIP-T3xG IP Phones device specification, which facilitates the discovery of a hardware component in a system, without the need for physical device configuration, or user intervention in resolving resource conflicts. ROM (Read-only Memory)--a class of storage medium used in computers and other electronic devices.
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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...
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Administrator’s Guide for SIP-T3xG 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 Luxembourg(Luxembourg) +01:00 Macedonia(Skopje)
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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...
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Administrator’s Guide for SIP-T3xG IP Phones This appendix describes the parameters you can set in the configuration files for the IP phone. The configuration files are <y0000000000xx>.cfg and <MAC>.cfg. You can set specific parameters in the configuration files for configuring the IP phones. The <y0000000000xx>.cfg and <MAC>.cfg files are stored on the provisioning server.
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Appendix Parameter- Configuration File network.internet_port.type <y0000000000xx>.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-...
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Administrator’s Guide for SIP-T3xG IP Phones Format IP Address Default Value Blank Range Not Applicable Example network.internet_port.mask = 255.255.255.0 Parameter- Configuration File network.internet_port.gateway <y0000000000xx>.cfg Configures the default gateway 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.
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Appendix Parameter- Configuration File network.secondary_dns <y0000000000xx>.cfg Configures the secondary 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...
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Administrator’s Guide for SIP-T3xG IP Phones effect. Format String Default Value Blank Range Not Applicable Example network.pppoe.user = xmyealink Parameter- Configuration File network.pppoe.password <y0000000000xx>.cfg Configures the PPPoE password 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.
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Appendix Parameter- Configuration File network.pc_port.ip <y0000000000xx>.cfg Configures the IP address for the PC port when the PC port is configured as Router. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value...
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Administrator’s Guide for SIP-T3xG IP Phones 0-Disabled 1-Enabled Example network.pc_port.dhcp_server = 1 Parameter- Configuration File network.dhcp.start_ip <y0000000000xx>.cfg Configure the start IP address that the IP phone assigns for the PC attached to the PC port when the PC port is configured as Router. Description Note: If you change this parameter, the IP phone will reboot to make the change take...
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Appendix Replace Rule Parameter- Configuration File dialplan.replace.prefix.x <y0000000000xx>.cfg Description Specifies the numbers you want to replace. Format String Default Value Blank Range Not Applicable Example dialplan.replace.prefix.1 = 123 Parameter- Configuration File dialplan.replace.replace.x <y0000000000xx>.cfg Specifies the alternate string instead of what Description the user enters.
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Administrator’s Guide for SIP-T3xG IP Phones 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.
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Appendix Format Integer Default Value Range Not Applicable Example phone_setting.dialnow_delay = 1 Area Code Parameter- Configuration File dialplan.area_code.code <y0000000000xx>.cfg Defines the area code to add before the Description entered numbers. Integer Format Default Value Blank Range Not Applicable Example dialplan.area_code.code = 010 Parameter- Configuration File dialplan.area_code.min_len...
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Administrator’s Guide for SIP-T3xG IP Phones 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 1 to 6 (for T38G)
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Appendix Valid values are: Range 1 to 6 (for T38G) 1 to 3 (for T32G) Example dialplan.block_out.line_id.1 = 1,2,3 Parameter- Configuration File phone_setting.active_backlight <y0000000000xx>.cfg _level Configures the backlight level used to adjust the backlight intensity of the LCD screen Description Level 1 is the least bright and level 10 is the most bright.
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Administrator’s Guide for SIP-T3xG IP Phones 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 set to 60 (60s), the LCD backlight is turned off when the IP phone is inactive for 60 seconds.
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Appendix Parameter- Configuration File security.user_password <y0000000000xx>.cfg Sets a new administrator password for the IP phone. The IP phone uses ―admin‖ as the default administrator password. Description Note: The IP phones support ASCII characters 32-126(0x20-0x7E) only in passwords. Format administrator username:new password Default Value admin Range...
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Administrator’s Guide for SIP-T3xG IP Phones Example phone_setting.lock = 2 Parameter- Configuration File phone_setting.phone_lock.unlo <y0000000000xx>.cfg ck_pin Sets a new unlock password. Once the IP phone is locked, you can use ―123‖ as the default password to unlock it. Description Note: The IP phones support numeric characters only in password.
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Appendix NTP Server Parameter- Configuration File local_time.manual_time_enable <y0000000000xx>.cfg Configures the phone to obtain the time and Description date manually or dynamically from the NTP server. Format Boolean Default Value Valid values are: Range 0-Manual 1-NTP Server Example local_time.manual_time_enable = 1 Parameter- Configuration File local_time.ntp_server1...
Administrator’s Guide for SIP-T3xG IP Phones ―local_time.manual_time_enable‖ is set to 1 (NTP Server). Format IP Address or Domain Name Default Value cn.pool.ntp.org Range Not Applicable Example local_time.ntp_server2 = 192.168.0.5 Parameter- Configuration File local_time.interval <y0000000000xx>.cfg Sets the IP phone to update time and date from the NTP server at regular intervals (in seconds).
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Appendix For more available time zone name list, refer Appendix B: Time Zones on page 221. Format String Default Value China(Beijing) Range Not Applicable Example local_time.time_zone_name = Korea(Seoul) Parameter- Configuration File local_time.summer_time <y0000000000xx>.cfg Enables or disables the use of Daylight Saving Description Time (DST).
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File local_time.start_time <y0000000000xx>.cfg Specifies the time to start DST. If ―local_time.dst_time_type‖ is set to 0 (By Date), use the mapping: MM: 1=Jan, 2=Feb,…, 12=Dec DD:1=the first day in a month,…, 31= the last day in a month HH:0=1am, 1=2am,…, 23=12pm If ―local_time.dst_time_type‖...
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Appendix If ―local_time.dst_time_type‖ is set to 1 (By Week), use the mapping: Month: 1=Jan, 2=Feb,…, 12=Dec Week of Month: 1=the first week in a month,…, 5=the last week in a month Day of Week: 1=Mon, 2=Tues,…, 7=Sun Hour of Day: 0=1am, 1=2am,…, 23=12pm Note: It works only if the parameter ―local_time.summer_time‖...
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Administrator’s Guide for SIP-T3xG IP Phones hour format. Format Integer Default Value Integer 0-12 Hour Range 1-24 Hour Example local_time.time_format = 0 Date Format Parameter- Configuration File local_time.date_format <y0000000000xx>.cfg Sets the date format. The IP phones support various date formats. Description You can change the date to your desired format according to your requirement.
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Appendix Format Default Value Blank Range Not Applicable The following example uses HTTP to download the language pack ―lang-Italian.txt‖(Italian) from the Example provisioning server 192.168.10.25. gui_lang.url = http://192.168.10.25/lang-Italian.txt Parameter- Configuration File lang.wui <y0000000000xx>.cfg Specifies the language used on the web user interface.
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Administrator’s Guide for SIP-T3xG IP Phones user interface. Format Text Default Value English Valid values are: English Chinese_S Chinese_T German Range French Italian Portuguese Polish Spanish Turkish Example lang.gui = Italian Parameter- Configuration File features.pound_key.mode <y0000000000xx>.cfg Defines the "#" or "*" key as the send key. If set to 0 (Disabled), neither ―#‖...
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Appendix Enables or disables the IP phone to play a tone when a user presses a send key. If set to 1 (Enabled), the IP phone plays a tone when a user presses a send key. Description Note: It works only if the key tone is enabled. So you should set the parameter ―features.key_tone‖...
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File features.hotline_delay <y0000000000xx>.cfg Specify the time (in seconds) the IP phone waits to automatically dial out the hotline number. If set to 0 (0s), the IP phone immediately dials out the preconfigured hotline number when you lift the handset, press the speakerphone Description key or press the line key.
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Appendix Parameter- Configuration File account.x.missed_calllog <MAC>.cfg Enables or disables the missed call log feature for account X. If 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.
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File call_waiting.enable <y0000000000xx>.cfg Enables or disables the call waiting feature. If set to 0 (Disabled), a new incoming call is automatically rejected by the IP phone with a busy message while during a call. Description If set to 1 (Enabled), the phone LCD screen presents a new incoming call while during a...
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Appendix Parameter- Configuration File auto_redial.enable <y0000000000xx>.cfg Enables or disables the IP phone to automatically redial the called number when it is busy. Description If set to 1 (Enabled), the IP phone dials the previous dialed out number automatically when the dialed number is busy. Format Boolean Default Value...
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Administrator’s Guide for SIP-T3xG IP Phones Default Value Range 1 to 300 Example auto_redial.times = 8 Parameter- Configuration File account.x.auto_answer <MAC>.cfg Enables or disables the auto answer feature for account X. If set to 1 (Enabled), the IP phone can automatically answer an incoming call.
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Appendix Format Boolean Default Value 0-Disabled Range 1-Enabled Example features.call_completion_enable = 1 Parameter- Configuration File account.x.anonymous_call <MAC>.cfg Enables or disables the anonymous call feature for account X. If set to 1 (Enabled), the IP phone blocks its identity from showing up to the callee when Description placing a call.
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File account.x.anonymous_call_offc <MAC>.cfg Sets the anonymous call off code to inform the server to disable the anonymous call feature for Description account X (optional). X ranges from 1 to 6. Format String Default Value Blank Range...
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Appendix (optional). X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable account.1.anonymous_reject_oncode = Example Parameter- Configuration File account.x.anonymous_reject_offc <MAC>.cfg Sets the anonymous call rejection off code to inform the server to disable the anonymous call rejection feature for account X Description (optional).
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File features.dnd.off_code <y0000000000xx>.cfg Sets the DND off code to inform the server to Description disable the DND feature (optional). Format String Default Value Blank Range Not Applicable Example features.dnd.off_code = *72 Parameter- Configuration File features.dnd.emergency_authori <y0000000000xx>.cfg...
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Appendix 404-No Found 480-Temporarily not available 486-Busy here Example features.dnd_refuse_code = 486 Parameter- Configuration File features.busy_tone_delay <y0000000000xx>.cfg Configure a period of time (in seconds) for which the busy tone is audible on the IP phone. When one party releases the call, a busy Description tone is audible to the other party indicating that the call connection breaks.
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Administrator’s Guide for SIP-T3xG IP Phones 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.
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Appendix 0-Disabled Range 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.
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Administrator’s Guide for SIP-T3xG IP Phones seconds) for account X. T4 represents the time the network will take to clear messages between the SIP Client and SIP Server. X ranges from 1 to 6. Format Float Default Value Example account.1.advanced.timer_t4 = 10 Parameter- Configuration File account.x.session_timer.enable...
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Appendix Range 1-999 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. If set to 0 (UAC), refreshing the session is performed by the IP phone. Description If set to 1 (UAS), refreshing the session is performed by a SIP server.
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File transfer.blind_tran_on_hook_ena <y0000000000xx>.cfg Enables or disables the IP phone to complete Description the blind transfer through on-hook. Format Boolean Default Value 0-Disabled Range 1-Enabled Example transfer.blind_tran_on_hook_enable = 1 Parameter- Configuration File transfer.on_hook_trans_enable <y0000000000xx>.cfg Enables or disables the IP phone to complete Description...
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Appendix Parameter- Configuration File account.x.conf_type <MAC>.cfg Defines the conference type for account X. If set to 0 (Local), conferences are set up on the IP phone locally. Description If set to 2 (Network Conference), conferences are set up by the server. X ranges from 1 to 6.
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File transfer.tran_others_after_conf_e <y0000000000xx>.cfg nable Enables or disables the Transfer on Conference Hang Up feature. If enabled, the other two parties remain connected when the conference initiator Description drops the conference call. Note: It is only applicable to the local conference.
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Appendix Parameter- Configuration File features.pickup.direct_pickup_co <y0000000000xx>.cfg Configures the direct pickup code on a phone basis. Description Note: The direct pickup code configured on a per-line basis takes precedence over that configured on a global basis. Format String Default Value Blank Range Not Applicable Example...
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Administrator’s Guide for SIP-T3xG IP Phones Format Boolean Default Value 0-Disabled Range 1-Enabled Example features.pickup.group_pickup_enable = 1 Parameter- Configuration File features.pickup.group_pickup_co <y0000000000xx>.cfg Configures the group pickup code on a phone basis. Description Note: The group pickup code configured on a per-line basis takes precedence over that configured on a global basis.
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Appendix Parameter- Configuration File account.x.dialoginfo_callpickup <MAC>.cfg Configures the Dialog-Info Call Pickup feature for account X. Description If set to 1 (Enabled), call pickup is implemented through SIP signals. X ranges from 1 to 6. Format Boolean Default Value 0-Disabled Range 1-Enabled Example account.1.dialoginfo_callpickup = 1...
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Administrator’s Guide for SIP-T3xG IP Phones Default Value Valid values are: 0-Disabled Range 1-HTTP & HTTPS 2-HTTP Only 3-HTTPS Only Example network.web_server_type = 2 Parameter- Configuration File network.port.http <y0000000000xx>.cfg Configures the HTTP port to access the web user interface of the IP phone. The default HTTP port is 80.
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Appendix Parameter- Configuration File account.x.cid_source <MAC>.cfg Configure 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‖ header. If the server does not send the ―PAI‖...
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Administrator’s Guide for SIP-T3xG IP Phones 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.
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Appendix 2-SIP INFO 3-AUTO+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 126 Example account.1.dtmf.dtmf_payload = 101 Parameter- Configuration File account.x.dtmf.info_type <MAC>.cfg...
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Administrator’s Guide for SIP-T3xG IP Phones Example features.dtmf.repetition = 2 Parameter- Configuration File features.intercom.allow <y0000000000xx>.cfg Enables or disables the IP phone to automatically answer an incoming intercom call. If set to 0 (Disabled), the IP phone rejects Description incoming intercom calls and sends a busy signal to the caller.
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Appendix Parameter- Configuration File features.intercom.tone <y0000000000xx>.cfg Enables or disables the IP phone to play a warning tone when receiving an intercom call. If set to 0 (Disabled), the IP phone Description automatically answers the intercom call without a warning tone. If set to 1 (Enabled), the IP phone plays a warning tone to alert you before answering the intercom call.
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File account.x.alert_info_url_enable <MAC>.cfg Enables or disables the distinctive ring tones feature for account X. Description X ranges from 1 to 6. Format Boolean Default Value 0-Enabled Range 1-Disabled Example account.1.alert_info_url_enable = 1 Parameter- Configuration File distinctive_ring_tones.alert_info.x.text...
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Appendix 1-Ring1.wav 2-Ring2.wav 3-Ring3.wav 4-Ring4.wav 5-Ring5.wav 6-Ring6.wav 7-Ring7.wav 8-Ring8.wav 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. Format Text Default Value Custom Valid values are: Custom Australia Austria Brazil ...
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Administrator’s Guide for SIP-T3xG IP Phones Portugal Spain Switzerland Sweden Russia United States Chile 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 voice.tone.info voice.tone.stutter voice.tone.message voice.tone.autoanswer Customizes the tone for each condition.
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Appendix Parameter- Configuration File features.remote_phonebook.ena <y0000000000xx>.cfg Enables or disables the IP phone to perform a Description remote phonebook search when receiving an incoming call. Format Boolean Default Value 0-Disabled Range 1-Enabled Example features.remote_phonebook.enable = 1 Parameter- Configuration File features.remote_phonebook.flas <y0000000000xx>.cfg h_time Sets how often to refresh the local cache of the remote phonebook.
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Administrator’s Guide for SIP-T3xG IP Phones prefix of the filter condition. Format String Default Value Blank 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 phone LCD screen.
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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 to login the LDAP server. It can be left blank in case the server Description allows anonymous to login. Otherwise you will need to provide the username to access the LDAP server.
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Administrator’s Guide for SIP-T3xG IP Phones cn=manager,dc=yealink,dc=cn Parameter- Configuration File ldap.password <y0000000000xx>.cfg Specifies the password to login the LDAP server. This parameter can be left blank in case the server allows anonymous to login. Description Otherwise you will need to provide the password to access the LDAP server.
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Appendix Format String 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.
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Administrator’s Guide for SIP-T3xG IP Phones Format Integer Default Value Range 2 or 3 Example ldap.version = 3 Parameter- Configuration File ldap.search_delay <y0000000000xx>.cfg Specifies the delay time (in milliseconds) to Description display the search results on the LCD screen after searching. Format Integer Default Value...
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Appendix 1-Enabled Example ldap.ldap_sort = 1 Parameter- Configuration File ldap.dial_lookup <y0000000000xx>.cfg Enables or disables the IP phone to perform Description an LDAP search when pre-dialing or dialing a call. Format Boolean Default Value 0-Disabled Range 1-Enabled Example ldap.dial_lookup = 1 Visual and Audio Alert for BLF Pickup Parameter- Configuration File...
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Administrator’s Guide for SIP-T3xG IP Phones Default Value 0-Disabled Range 1-Enabled Example features.pickup.blf_audio_enable = 1 Parameter- Configuration File account.x.blf.blf_list_uri <MAC>.cfg Specifies the URI used to access the BLF list configured on the SIP server for account X. Description X ranges from 1 to 6. Format String Default Value...
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Appendix X ranges from 1 to 6. Format Integer Default Value Valid values are: 0-Disabled Range 1-Broadsoft SCA 2-BLA Example account.1.shared_line = 1 Parameter- Configuration File account.x.enable <MAC>.cfg Enables or disables account X. Description X ranges from 1 to 6. Format Boolean Default Value...
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Administrator’s Guide for SIP-T3xG IP Phones Format String Default Value Blank Range Not Applicable Example account.1.display_name = 2413333601 Parameter- Configuration File account.x.auth_name <MAC>.cfg Configures the register name of the account X ranges from 1 to 6. Description Note: If configuring the secondary line on the IP phone, enter the register name of the primary line for this parameter.
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Appendix Range Not Applicable Example account.1.user_name = 2413333601 Parameter- Configuration File account.x.sip_server_host <MAC>.cfg Configures the SIP server address for account X. Description X ranges from 1 to 6. Format IP Address or Domain Name Default Value Blank Range Not Applicable account.1.sip_server_host = Example server@domain name.net...
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File account.x.outbound_host <MAC>.cfg Configures the address of the outbound proxy server for account X. Description X ranges from 1 to 6. Format IP Address Default Value Blank Range Not Applicable Example account.1.outbound_host = 199.19.195.10 Parameter- Configuration File account.x.outbound_port...
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Appendix Format Boolean Default Value 0-Disabled Range 1-Enabled Example bw.feature_key_sync = 1 Parameter- Configuration File account.x.subscribe_mwi <MAC>.cfg Enables or disables the IP phone to subscribe the message waiting indicator for account X. Description If set to 1 (Disabled), the IP phone sends a SUBSCRIBE message to the server for message-summary updates.
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Administrator’s Guide for SIP-T3xG IP Phones Format Integer Default Value 3600 Value 0 to 84600 Example account.1.subscribe_mwi_expires = 3600 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 action_url.incoming_call action_url.outgoing_call action_url.call_established action_url.dnd_on action_url.dnd_off action_url.always_fwd_on action_url.always_fwd_off action_url.busy_fwd_on action_url.busy_fwd_off action_url.no_answer_fwd_on action_url.no_answer_fwd_off action_url.transfer_call action_url.blind_transfer_call action_url.attended_transfer_call...
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Appendix 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://IP address of server/help.xml? variable name=variable value Valid variable values are: $mac $model $firmware Description $active_url $active_user ...
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Administrator’s Guide for SIP-T3xG IP Phones handles HTTP GET requests from any IP address. Format IP Address Default Value Blank Range IP address or any Example features.action_uri_limit_ip = any Parameter- Configuration File account.x.transport <MAC>.cfg Configures the transport type for account X. If set to 3 (DNS SRV), the IP phone is able to perform DNS SRV query, and fail over the Description...
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Appendix Format Boolean 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 transmission of LLDP packets. Note: If you change this parameter, the IP Description phone will reboot to make the change take effect.
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Administrator’s Guide for SIP-T3xG 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. Format Integer Default Value...
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Appendix 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 Range...
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Administrator’s Guide for SIP-T3xG IP Phones Format Boolean Default Value 0-Disabled Range 1-Enabled Example network.vpn_enable = 1 Parameter- Configuration File openvpn.url <y0000000000xx>.cfg 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...
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Appendix The default DSCP value for SIP packets is 26 (Assured Forwarding). Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 0 to 63 Range Example network.qos.signaltos = 30 Parameter- Configuration File account.x.nat.nat_traversal...
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File account.x.nat.stun_port <MAC>.cfg Specifies the port of the STUN server. Description X ranges from 1 to 6. Format Integer Default Value 3478 Range Not Applicable Example account.1.nat.stun_port = 3479 Parameter- Configuration File network.802_1x.mode <y0000000000xx>.cfg Specifies the types of the 802.1X...
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Appendix Parameter- Configuration File network.802_1x.md5_password <y0000000000xx>.cfg Enters the password used for authenticating the IP phone. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format String Default Value Blank Range Not Applicable network.802_1x.md5_password = Example admin123...
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Administrator’s Guide for SIP-T3xG IP Phones When Y=11, the default value is 0; When Y=12, the default value is 0; When Y=13, the default value is 0. 0-Disabled Range 1-Enabled Example account.1.codec.1.enable = 1 Parameter- Configuration File account.X.codec.Y.payload_type <MAC>.cfg Specifies the codec for account X to use. Description X ranges from 1 to 6.
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Appendix 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. Y ranges from 1 to 13. Format Integer When Y=1, the default value is 1; When Y=2, the default value is 2; When Y=3, the default value is 4;...
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Administrator’s Guide for SIP-T3xG IP Phones When Y=4, the default value is 4; When Y=5, the default value is 18; When Y=6, the default value is 9; When Y=7, the default value is 102; When Y=8, the default value is 112; When Y=9, the default value is 102;...
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Appendix Example voice.echo_cancellation = 1 Parameter- Configuration File voice.vad <y0000000000xx>.cfg Enables or disables the VAD feature on the Description IP phone. Format Boolean Default Value 0-Disabled Range 1-Enabled Example voice.vad = 1 Parameter- Configuration File voice.cng <y0000000000xx>.cfg Enables or disables the CNG feature on the Description IP phone.
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Administrator’s Guide for SIP-T3xG IP Phones 1-Adaptive Example voice.jib.adaptive = 1 Parameter- Configuration File voice.jib.min <y0000000000xx>.cfg Configures the minimum delay time (in milliseconds) for jitter buffer. Description Note: It works only if the parameter ―voice.jib.adaptive‖ is set to 1 (Adaptive). Format Integer Default Value...
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Appendix Example voice.jib.mormal = 100 Parameter- Configuration File account.x.transport <MAC>.cfg Configures the transport type for account X. If set to 2 (TLS), the SIP message of this Description account will be encrypted after the successful TLS negotiation. X ranges from 1 to 6. Format Integer Default Value...
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File account.x.enable_signal_encode <MAC>.cfg Enables or disables the IP phone to encode SIP signal using RC4 encryption algorithm. Description X ranges from 1 to 6. Format Boolean Default Value Valid values are: Value 0-Disabled 1-Enabled Example...
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Appendix http://192.168.1.20/tc.crt Parameter- Configuration File server_certificates.url <y0000000000xx>.cfg Specifies the access URL of the server certificate. Description Note: The server certificate you want to add must have a .pem extension. Format String Default Value Blank Range Not Applicable server_certificates.url = Example http://192.168.1.20/ca.pem Parameter- Configuration File...
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File auto_provision.aes_key_16.com <y0000000000xx>.cfg Configures the AES key which is used to Description encrypt or decrypt the <y0000000000xx>.cfg file. String Format () ><| "& cannot be included. Default Value Blank Range 16 characters auto_provision.aes_key_16.com = Example 0123456789abcdef...
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Appendix Valid values are: Range 0-Disabled 1-Enabled Example auto_provision.mode = 1 Parameter- Configuration File auto_provision.repeat.enable <y000000000068>.cfg Description Enable or disable the IP phone to check the new configuration repeatedly. Format Boolean Default Value 0-Disabled Range 1-Enabled Example auto_provision.repeat.enable = 1 Parameter- Configuration File auto_provision.repeat.minutes...
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Administrator’s Guide for SIP-T3xG IP Phones 0-Disabled Range 1-Enabled Example auto_provision.weekly.enable = 1 Parameter- Configuration File auto_provision.weekly.mask <y000000000068>.cfg Description Defines the desired day(s) of a week for the phone to check new configuration. Note: It works only if the parameter ―auto_provision.weekly.enable‖...
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Appendix Parameter- Configuration File auto_provision.weekly.end_time <y000000000068>.cfg Description Sets the end time of day in 24-hour period for the phone to check new configuration files. Note: It works only if the parameter ―auto_provision.weekly.enable‖ is set to 1 (Enabled). Format 00:00 Default Value 00:00 Range 00:00 to 23:59...
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Administrator’s Guide for SIP-T3xG IP Phones http://192.168.10.25/dialplan.xml Parameter- Configuration File dialplan_dialnow.url <y0000000000xx>.cfg Specifies the access URL of the dial-now Description template. Format Default Value Blank Range Not Applicable dialplan_dialnow.url = Example http://192.168.10.25/dialnow.xml Parameter- Configuration File wallpaper_upload.url <y0000000000xx>.cfg Specifies the access URL of the wallpaper Description image.
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Appendix Range Not Applicable screen_saver.pic.url = Example http://192.168.10.25/screensaver.jpg Parameter- Configuration File custom_softkey_call_failed.url <y0000000000xx>.cfg Specifies the access URL of the customized file for the soft key presented on the phone Description LCD screen when in the CallFailed state. Format Default Value Not Applicable Range Not Applicable...
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File custom_softkey_connecting.url <y0000000000xx>.cfg Specifies the access URL of the customized file Description for the soft key presented on the phone LCD screen when in the Connecting state. Format Default Value Not Applicable Range Not Applicable The following example uses HTTP to...
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Appendix Default Value Not Applicable Range Not Applicable The following example uses HTTP to download the RingBack state file from the ―XMLfiles‖ directory on provisioning server 10.2.8.16 using 8080 port. Example custom_softkey_ring_back.url = http://10.2.8.16:8080/XMLfiles/RingBack.x 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 phone LCD screen when in the Talking state.
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File remote_phonebook.data.x.url <y0000000000xx>.cfg Specifies the access URL of the remote XML phonebook. Description X ranges from 1 to 5. Format Default Value Blank Range Not Applicable remote_phonebook.data.1.url = Example http://192.168.1.20/phonebook.xml Parameter- Configuration File syslog.server <y0000000000xx>.cfg Specifies the IP address of the syslog...
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Appendix effect. Format Integer Default Value Range 0 to 6 Example syslog.log_level = 2 Parameter- Configuration File watch_dog.enable <y0000000000xx>.cfg Description Enables or disables the Watch Dog feature. Format Boolean Default Value 0-Disabled Range 1-Enabled Example watch_dog.enable = 1 This section provides the DSS key parameters you can configure on the IP phone. The DSS key is consist of the memory key and line key.
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Administrator’s Guide for SIP-T3xG IP Phones Various key features can be assigned to the DSS key. The SIP-T38G IP phone supports to be assigned features to the memory keys and line keys. The SIP-T32G IP phone only supports to be assigned features to the line keys. The configurations of the line key are basically the same as the memory key.
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Appendix Hot Desking Zero Touch Keypad Lock Format Integer Default Value 0 (Auto) Valid values are: Range 0 to 6 (for T38G) 0 to 3 (for T32G) Example memorykey.1.line = 2 Parameter- Configuration File memorykey.x.value <y0000000000xx>.cfg Specifies the value for some key features.
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Specifies the key feature for the memory key. X ranges from 1 to 10. Valid types are: N/A (default for memory key) Conference Forward Transfer Hold ...
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Appendix Format Integer Default Value 0 (N/A) Valid values are: 0-N/A(default for memory key) 1-Conference 2-Forward 3-Transfer 4-Hold 5-DND 6-Redial 7-Call Return 8-SMS 9-Call Pickup 10-Call Park 11-DTMF 12-Voicemail 13-SpeedDial 14-Intercom 15-Line(default for line key) Range 16-BLF 17-URL 18-Group Listening 19-Public Hold 20-Private Hold 21- Shared Line...
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Administrator’s Guide for SIP-T3xG IP Phones 45-Local Group 46-BroadSoft Group 50-Keypad Lock Example memorykey.1.type = 8 Parameter- Configuration File memorykey.x.xml_phonebook <y0000000000xx>.cfg Specifies the desired phonebook when multiple phonebooks are configured on the IP phone. Description This parameter only applies to the Local Group/XML Group/BroadSoft Group features.
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Appendix DND Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Configures a memory key to be DND key on the IP phone. Description The digit 5 stands for the key type DND. X ranges from 1 to 10. Format Integer Value Example memorykey.1.type = 5 Direct Pickup Key Parameter-...
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File memorykey.x.value <y0000000000xx>.cfg Specifies the direct pickup feature code followed by the number of monitored Description extension. X ranges from 1 to 10. Format String Range Not Applicable Example memorykey.1.value = *971001 Group Pickup Key Parameter- Configuration File...
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Appendix Parameter- Configuration File memorykey.x.value <y0000000000xx>.cfg Specifies the group pickup feature code. Description X ranges from 1 to 10. Format String Range Not Applicable Example memorykey.1.value = *98 Call Return Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Configures a memory key to be call return key on the IP phone.
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Administrator’s Guide for SIP-T3xG IP Phones Parameter- Configuration File memorykey.x.line <y0000000000xx>.cfg Specifies the desired line to apply the call park key. Description X ranges from 1 to 10. Format Integer Valid values are: Range 0 to 5 (for T38G) 0 to 2 (for T32G) Example memorykey.2.line = 0 Parameter-...
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Appendix Parameter- Configuration File memorykey.x.line <y0000000000xx>.cfg Specifies the desired line to apply the intercom key. Description X ranges from 1 to 10. Format Integer Valid values are: Range 0 to 6 (for T38G) 0 to 3 (for T32G) Example memorykey.2.line = 1 Parameter- Configuration File memorykey.x.value...
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Administrator’s Guide for SIP-T3xG IP Phones BLF Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Configures a memory key to be BLF key on the IP phone. Description The digit 16 stands for the key type BLF. X ranges from 1 to 10. Format Integer Value...
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Appendix feature. This parameter only applies to the BLF feature. X ranges from 1 to 10. Format String Default Value Blank Range Not Applicable Example memorykey.3.pickup_value = *88 BLF List Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Configures a memory key to be BLF list key on the IP phone.
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Administrator’s Guide for SIP-T3xG IP Phones Shared Line Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Configures a memory key to be a shared line key on the IP phone. Description The digit 21 stands for the key type Shared Line. X ranges from 1 to 10. Format Integer Value...
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Appendix ACD Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Configures a memory key to be an ACD key on the IP phone. Description The digit 42 stands for the key type ACD. X ranges from 1 to 10. Format Integer Value Example memorykey.2.type = 42 Parameter-...
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Administrator’s Guide for SIP-T3xG IP Phones URL Record Key Parameter- Configuration File memorykey.x.type <y0000000000xx>.cfg Configures a memory key to be a URL record key on the IP phone. Description The digit 35 stands for the key type URL Record. X ranges from 1 to 10. Format Integer Value...
Appendix This section describes how the Yealink SIP-T3xG 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 ...
RFC number. The following SIP request messages are supported: Method Supported Notes REGISTER The Yealink SIP-T3xG IP phones support mid-call changes such as putting a INVITE call on hold as signaled by a new INVITE that contains...
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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...
Administrator’s Guide for SIP-T3xG IP Phones Method Supported Notes Expires From Max-Forwards Min-SE P-Asserted-Identity P-Preferred-Identity Proxy-Authenticate Proxy-Authorization RAck Record-Route Refer-To Referred-By Remote-Party-ID Replaces Require Route RSeq Session-Expires Subscription-State Supported User-Agent The following SIP responses are supported: 1xx Response—Information Responses 1xx Response Supported Notes 100 Trying...
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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 202 Accepted In REFER transfer. 3xx Response—Redirection Responses 3xx Response Supported Notes 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 4xx Response—Request Failure Responses 4xx Response...
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Administrator’s Guide for SIP-T3xG 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 481 Call/Transaction Does Not Exist 482 Loop Detected...
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Appendix 5xx Response—Server Failure Responses 5xx Response Supported Notes 500 Internal Server Error 501 Not Implemented 502 Bad Gateway 503 Service Unavailable 504 Gateway Timeout 505 Version Not Supported 6xx Response—Global Responses 6xx Response Supported Notes 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable SDP Headers...
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Administrator’s Guide for SIP-T3xG 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. ...
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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 the Yealink SIP IP phones. The call flow scenario is as follows: User A calls User B.
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Administrator’s Guide for SIP-T3xG 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 ...
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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 the Yealink SIP IP phones.
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Administrator’s Guide for SIP-T3xG 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. User A Proxy Server User B...
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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 ...
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Administrator’s Guide for SIP-T3xG 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.
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In this scenario, the two end users are User A and User B. User A and User B are located at the Yealink SIP IP phones. The call flow scenario is as follows: User A calls User B.
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Administrator’s Guide for SIP-T3xG 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 ...
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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.
Administrator’s Guide for SIP-T3xG 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 the Yealink SIP IP phones.
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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 ...
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200 OK response. The following figure illustrates a successful call between Yealink SIP IP phones in which parties are in a call, one of the participants receives a call from a third party, then...
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Appendix and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: User A calls User B. User B answers the call. User C calls User B.
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Administrator’s Guide for SIP-T3xG 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 ...
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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.
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Administrator’s Guide for SIP-T3xG 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.
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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.
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Administrator’s Guide for SIP-T3xG 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.
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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 ...
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Administrator’s Guide for SIP-T3xG 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. The proxy server sends the SIP ACK to User B.
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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.
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Administrator’s Guide for SIP-T3xG 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.
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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 ...
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Administrator’s Guide for SIP-T3xG 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. The proxy server sends the SIP ACK to User B.
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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.
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Administrator’s Guide for SIP-T3xG 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.
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.
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Administrator’s Guide for SIP-T3xG 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 ...
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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.
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.
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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 ...
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Administrator’s Guide for SIP-T3xG 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 proxy server that User A has received the ACK message.
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.
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Administrator’s Guide for SIP-T3xG 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 ...
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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.
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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.
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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.
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Administrator’s Guide for SIP-T3xG 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 ...
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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.
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Administrator’s Guide for SIP-T3xG 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.
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.
Administrator’s Guide for SIP-T3xG IP Phones dialplan.replace.prefix.1 = dialplan.replace.replace.1 = dialplan.replace.line_id.1 = Time Settings local_time.time_zone = local_time.time_zone_name = local_time.ntp_server1 = local_time.ntp_server2 = local_time.interval = #Use the following parameters to set the time and date manually. local_time.manual_time_enable = local_time.date_format = local_time.time_format = Auto DST Settings local_time.summer_time = local_time.dst_time_type =...
Index Numeric 180 Ring Workaround Call Completion 802.1x Authentication Call Forward Call Hold Call Log Call Park/Retrieve 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...
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Administrator’s Guide for SIP-T3xG IP Phones Network Conference No Answer Forward Getting Information from Status Indicators Getting Started Group Pickup Phone Lock Phone User Interface Physical Features of the SIP-T3xG IP Phones H.323 Hot Desking Product Overview Hotline Quality of Service In This Guide Index Initialization Process Overview...
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Index Table of Contents Time and Date Transfer on Conference Hang Up Transport Layer Security (TLS) Troubleshooting Troubleshooting Methods Troubleshooting Solutions Upgrading Firmware Use Outbound Proxy in Dialog User Agent Client (UAC) User Agent Server (UAS) User Password Verifying Startup Viewing Log Files VLAN Voice Activity Detection...
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