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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.
Yealink IP phone firmware contains third-party software under the GNU General Public License (GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms and conditions of the license. The original GPL license, source code of components licensed under GPL and used in Yealink products can be downloaded from Yealink web site: http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293.
About This Guide This guide is intended for administrators who need to properly configure, customize, manage, and troubleshoot the IP phone system rather than end-users. It provides details on the functionality and configuration of IP phones. Many of the features described in this guide involve network settings, which could affect the IP phone’s performance in the network. So an understanding of IP networking and a prior knowledge of IP telephony concepts are necessary.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Chapter 4, “Configuring Advanced Features” describes how to configure the advanced features on IP phones. Chapter 5, “Configuring Audio Features” describes how to configure the audio features on IP phones. Chapter 6, “Configuring Security Features” describes how to configure the security features on IP phones. Chapter 7, “Upgrading Firmware” describes how to upgrade firmware of IP phones.
About This Guide Upgrading Firmware on page 225 Resource Files on page 229 Documentations of the newly released SIP-T19P and SIP-T21P IP phones have also been added.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The following sections are new for this version: Hot Desking on page 162 TR-069 Device Management on page 194 IPv6 Support on page 196 Major updates have occurred to the following sections: Configuring Network Parameters Manually on page 24 Softkey Layout on page 61 Directed Call Pickup on page 107 Distinctive Ring Tones on page 131 Automatic Call Distribution on page 149 Action URL on page 166 Server Redundanc
About This Guide Return Code When Refuse on page 89 Early Media on page 90 180 Ring Workaround on page 90 Use Outbound Proxy in Dialog on page 92 SIP Session Timer on page 93 Session Timer on page 94 Call Return on page 115 Transfer via DTMF on page 125 Intercom on page 126 Music on Hold on page 148 Automatic Call Distribution on page 149 Message Waiting Indicator on page 151 Multicast Paging on page 153 Call Recording on page 158 LLDP on page 176
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Hot Desking on page 162 Action URL on page 166 Action URI on page 166 Resource Files on page 229 Appendix C: Configuration Parameters on page 260 Appendix F: Sample Configuration File on page 470 Major updates have occurred to the following sections: x Creating Dial Plan on page 32 Phone Lock on page 48 Time and Date on page 50 Busy Lamp Field on page 142
Table of Contents About This Guide ...................................................................... v Documentations ............................................................................................................................... v In This Guide .................................................................................................................................... v Summary of Changes ..........................................................................................
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Configuring Transmission Methods of the Internet Port and PC Port ................................. 28 Configuring PC Port Mode ..................................................................................................... 30 Creating Dial Plan ......................................................................................................................... 32 Replace Rule ...............................................................
Table of Contents Network Conference ................................................................................................................... 105 Transfer on Conference Hang Up .............................................................................................. 106 Directed Call Pickup .................................................................................................................... 107 Group Call Pickup.................................................................
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Configuring Audio Features ..................................................199 Headset Prior ............................................................................................................................... 199 Dual Headset ............................................................................................................................... 200 Audio Codecs ....................................................................
Table of Contents Why doesn’t the IP phone display time and date correctly? ........................................... 248 Why do I get poor sound quality during a call? ................................................................ 248 What is the difference between a remote phone book and a local phone book? ....... 249 What is the difference among user name, register name and display name? ............. 249 How to reboot the IP phone remotely? .........................................................
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Call Transfer without Consultation ...................................................................................... 446 Call Transfer with Consultation ............................................................................................ 450 Always Call Forward ............................................................................................................ 455 Busy Call Forward ...............................................
Product Overview This chapter contains the following information about IP phones: VoIP Principle SIP Components SIP IP Phone Models VoIP VoIP (Voice over Internet Protocol) is a technology using the Internet Protocol instead of traditional Public Switch Telephone Network (PSTN) technology for voice communications. It is a family of technologies, methodologies, communication protocols, and transmission techniques for the delivery of voice communications and multimedia sessions over IP networks.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones SIP provides capabilities to: Determine the location of the target endpoint -- SIP supports address resolution, name mapping, and call redirection. Determine media capabilities of the target endpoint -- Via Session Description Protocol (SDP), SIP determines the “lowest level” of common services between endpoints. Conferences are established using only media capabilities that can be supported by all endpoints.
Product Overview preferential to use this method when not using an application layer firewall. Application layer firewalls like to know what applications are flowing though which ports and it is possible to use content types of other applications other than the one you are trying to let through what has been denied. User agent server (UAS) UAS is a server that hosts the application responsible for receiving the SIP requests from a UAC, and on reception it returns a response to the request back to the UAC.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Routers are configured for VoIP. VoIP gateways are configured for SIP. The latest (or compatible) firmware of IP phones is available. A call server is active and configured to receive and send SIP messages. This section lists the available physical features of IP phones.
Product Overview SIP-T26P Physical Features: - TI TITAN chipset and TI voice engine - 132x64 graphic LCD - 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated - HD Voice: HD Codec, HD Handset, HD Speaker - 45 keys including 13 DSS keys - 1xRJ9 (4P4C) handset port - 1xRJ9 (4P4C) headset port - 2xRJ45 10/100Mbps Ethernet ports - 1XRJ12 (6P6C) expansion module port - 16 LEDs: 1xpower, 3xline, 1xmessage, 1xheadset, 10xmemory - Power adapter: AC 100~240V input and DC 5V/1.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones SIP-T22P Physical Features: 6 - TI TITAN chipset and TI voice engine - 132x64 graphic LCD - 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated - HD Voice: HD Codec, HD Handset, HD Speaker - 32 keys including 4 soft keys - 1xRJ9 (4P4C) handset port - 1xRJ9 (4P4C) headset port - 2xRJ45 10/100Mbps Ethernet ports - 5 LEDs: 1xpower, 3xline, 1xmessage - Power adapter: AC 100~240V input and DC 5V/1.
Product Overview SIP-T21P Physical Features: - 132x64 graphic LCD - 2 VoIP accounts - 31 keys including 4 soft keys - 4 LEDs: 1xpower, 2xline, 1xmessage - HD Voice: HD Codec, HD Handset, HD Speaker - 1xRJ9 (4P4C) handset port - 1xRJ9 (4P4C) headset port - 2xRJ45 10/100Mbps Ethernet ports - Power adapter: AC 100~240V input and DC 5V/600mA output - Power over Ethernet (IEEE 802.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones SIP-T20P Physical Features: 8 - TI TITAN chipset and TI voice engine - 3-line LCD consists of an icon line and two 15-character lines - 2 VoIP accounts, BroadSoft/Avaya/Asterisk validated - HD Voice: HD Codec, HD Handset, HD Speaker - 31 keys including 9 function keys - 1xRJ9 (4P4C) handset port - 1xRJ9 (4P4C) headset port - 2xRJ45 10/100Mbps Ethernet ports - 4 LEDs: 1xpower, 2xline, 1xmessage - Power adapter: AC 100~240V input
Product Overview SIP-T19P Physical Features: - 132x64 graphic LCD - Single VoIP account - 29 keys including 4 soft keys - 1xRJ9 (4P4C) handset port - 1xRJ9 (4P4C) headset port - 2xRJ45 10/100Mbps Ethernet ports - 1 LED: 1xpower - Power adapter: AC 100~240V input and DC 5V/600mA output - Power over Ethernet (IEEE 802.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones In addition to physical features introduced above, IP phones also support the following key features when running the latest firmware: Phone Features - Call Options: emergency call, call waiting, call hold, call mute, call forward, call transfer, call pickup, conference. - Basic Features: DND, phone lock, auto redial, live dialpad, dial plan, hotline, caller identity, auto answer.
Product Overview - Dial URL via SIP server - TR-069 Security - HTTPS (server/client) - SRTP (RFC3711) - Transport Layer Security (TLS) - VLAN (802.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 12
Getting Started This chapter provides basic information and installation instructions of IP phones. This chapter provides the following sections: Connecting the IP Phones Initialization Process Overview Verifying Startup Configuration Methods Reading Icons Configuring Basic Network Parameters Creating Dial Plan This section introduces how to install IP phones with components in packaging contents. Note 1. Attach the stand 2. Connect the handset and optional headset 3.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 1) Attach the stand: SIP-T28P/T26P SIP-T22P/T21P/T20P SIP-T19P 14
Getting Started 2) Connect the handset and optional headset: SIP-T28P/T26P SIP-T22P/T21P/T20P/T19P 3) Connect the network and power: AC power Power over Ethernet (PoE) AC Power To connect the AC power and network: 1. Connect the DC plug of the power adapter to the DC5V port on the IP phone and connect the other end of the power adapter into an electrical power outlet. 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Power over Ethernet With the included or a regular Ethernet cable, IP phones can be powered from a PoE-compliant switch or hub. To connect the PoE: 1. Connect the Ethernet cable between the Internet port on the IP phone and an available port on the in-line power switch/hub. Note If in-line power switch/hub is provided, you don’t need to connect the phone to the power adapter. Make sure the switch/hub is PoE-compliant.
Getting Started Querying the DHCP (Dynamic Host Configuration Protocol) Server The IP phone is capable of querying a DHCP server. DHCP is enabled on the IP phone by default. The following network parameters can be obtained from the DHCP server during initialization: IP Address Subnet Mask Gateway Primary DNS (Domain Name Server) Secondary DNS You need to configure network parameters of the IP phone manually if any of them is not supplied by the DHCP server.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. The message “Initializing, Please Wait” appears on the LCD screen when the IP phone starts up. 3. 4. The main LCD screen displays the following: Time and date Soft key labels (not applicable to SIP-T20P IP phones) Press the OK key to check the IP phone status, the LCD screen displays the valid IP address, MAC address, firmware version, etc.
Getting Started CFG file is named after the MAC address of the IP phone. For example, if the MAC address of a SIP-T22P IP phone is 001565113af8, names of these two configuration files must be: y000000000005.cfg and 001565113af8.cfg. The name of the Common CFG file for each IP phone model is: SIP-T28P: y000000000000.cfg SIP-T26P: y000000000004.cfg SIP-T22P: y000000000005.cfg SIP-T21P: y000000000034.cfg SIP-T20P: y000000000007.cfg SIP-T19P: y000000000031.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The latest values configured on the IP phone take effect finally. Icons associated with different features may appear on the LCD screen. The following table provides a description for each icon on IP phones.
Getting Started SIP-T28P SIP-T26P SIP-T22P SIP-T21P SIP-T20P SIP-T19P Description Ringer volume is / 0 Phone Lock Received Calls Placed Calls Missed Calls / / / / / / Recording box is full A call cannot be recorded Recording starts successfully Recording / / cannot be started Recording / / cannot be stopped This section describes how to configure basic network parameters for the IP phone. Note This section mainly introduces IPv4 network parameters. IP phones also support IPv6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones be configured and used when DHCP is enabled. DHCP Option DHCP provides a framework for passing information to TCP/IP network devices. Network and other control information are carried in tagged data items that are stored in the options field of the DHCP message. The data items themselves are also called options. DHCP can be initiated by simply connecting the IP phone with the network.
Getting Started Parameter DHCP Option Name Description options. Identify a boot file when the 'file' field in the Boot file Name 67 DHCP header has been used for DHCP options. Procedure DHCP can be configured using the configuration files or locally. Configure DHCP on the IP phone. Configure static DNS address Configuration File .cfg when DHCP is used. For more information, refer to DHCP on page 260. Configure DHCP on the IP phone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 3. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. 4. Click OK to reboot the IP phone. To configure static DNS address when DHCP is used via web user interface: 1. Click on Network->Basic. 2. In the IPv4 Config block, mark the DHCP radio box. 3. Mark the Static DNS radio box. 4. Enter the desired values in the Primary DNS and Secondary DNS fields. 5.
Getting Started Default Gateway Primary DNS Secondary DNS Procedure Network parameters can be configured manually using the configuration files or locally. Configure network parameters of the IP phone manually. Configuration File .cfg For more information, refer to Static Network Settings on page 261. Configure network parameters of the IP phone manually.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure a static IPv4 address via web user interface: 1. Click on Network->Basic. 2. In the IPv4 Config block, mark the Static IP Address radio box. 3. Enter the desired values in the IP Address, Subnet Mask, Gateway, Primary DNS and Secondary DNS fields. 4. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. 5. Click OK to reboot the IP phone.
Getting Started PPPoE (Point-to-Point Protocol over Ethernet) is a network protocol used by Internet Service Providers (ISPs) to provide Digital Subscriber Line (DSL) high speed Internet services. PPPoE allows an office or building-full of users to share a common DSL connection to the Internet. PPPoE connection is supported by the IP phone Internet port. Contact your ISP for the PPPoE user name and password. Procedure PPPoE can be configured using the configuration files or locally.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 5. Click OK to reboot the IP phone. To configure PPPoE via phone user interface: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN Port->IPv4->PPPoE IP Client. 2. Enter the user name and password in corresponding fields. 3. Press the Save soft key to accept the change. The IP phone reboots automatically to make settings effective after a period of time.
Getting Started Half-duplex Half-duplex transmission refers to transmitting voice or data in both directions, but in one direction at a time; this means one device can send data on the line, but not receive data simultaneously. You can configure the half-duplex transmission on both Internet port and PC port for the IP phone to transmit in 10Mbps or 100Mbps.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Configure the transmission methods of Ethernet ports. Local Web User Interface Navigate to: http:///servlet ?p=network-adv&q=load To configure the transmission methods of Ethernet ports via web user interface: 1. Click on Network->Advanced. 2. Select the desired value from the pull-down list of WAN Port Link. 3. Select the desired value from the pull-down list of PC Port Link. 4. Click Confirm to accept the change.
Getting Started Procedure PC port mode can be configured using the configuration files or locally. Configure the PC port mode. Configuration File .cfg For more information, refer to PC Port Mode on page 266. Configure the PC port mode. Web User Interface Local Navigate to: http:///servlet ?p=network-pcport&q=load Phone User Interface Configure the PC port mode. To configure the PC port mode via web user interface: 1. Click on Network->PC Port. 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure the PC port mode via phone user interface: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->PC Port. 2. Select the desired mode. If you select Router, you can configure the IP address for the PC port and configure DHCP for the PC attached to the PC port. 1) Enter the IP address in the IPv4 field. 2) Enter the subnet mask in the Subnet Mask field.
Getting Started The dash “-” can be used to match a range of characters within the - brackets. Example: “[5-7]” would match the number “5”, ”6” or ”7”. The comma “,” can be used as a separator within the bracket. , Example: “[2,5,8]” would match the number ”2”, “5” or “8”. The square bracket "[]" can be used as a placeholder for a single [] character which matches any of a set of characters. Example: "91[5-7]1234"would match “9151234”, “9161234”, “9171234”.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To create a replace rule via web user interface: 1. Click on Settings->Dial Plan->Replace Rule. 2. Enter the string in the Prefix field. 3. Enter the string in the Replace field. 4. Enter the desired line ID in the Account field or leave it blank. If you leave this field blank or enter 0, the replace rule will apply to all accounts on the IP phone. 5. Click Add to add the replace rule.
Getting Started dial-now rule. For more information, refer to Dial Plan on page 269. Create the dial-now rule for the IP phone. Navigate to: http:///servlet Local Web User Interface ?p=settings-dialnow&q=load Configure the delay time for the dial-now rule. Navigate to: http:///servlet ?p=features-general&q=load To create a dial-now rule via web user interface: 1. Click on Settings->Dial Plan->Dial-now. 2. Enter the desired value in the Rule field. 3.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. Enter the desired time within 1-14 (in seconds) in the Time-Out for Dial-Now Rule field. 3. 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 before the numbers when dialing out them. IP phones only support one area code rule.
Getting Started ?p=settings-areacode&q=load To configure an area code rule via web user interface: 1. Click on Settings->Dial Plan->Area Code. 2. Enter the desired values in the Code, Min Length (1-15) and Max Length (1-15) fields. 3. Enter the desired line ID in the Account field or leave it blank. If you leave this field blank or enter 0, the area code rule will apply to all accounts on the IP phone. 4. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones ?p=settings-blackout&q=load To create a block out rule via web user interface: 1. Click on Settings->Dial Plan->Block Out. 2. Enter the desired value in the BlockOut Number field. 3. Enter the desired line ID in the Account field or leave it blank. If you leave this field blank or enter 0, the block out rule will apply to all accounts on the IP phone. 4. 38 Click Confirm to add the block out rule.
Configuring Basic Features This chapter provides information for making configuration changes for the following basic features: Power Indicator LED Contrast Backlight User Password Administrator Password Phone Lock Time and Date Language Logo Customization Softkey Layout Key as Send Hotline Call Log Missed Call Log Local Directory Live Dialpad Call Waiting Auto Redial Auto Answer Call Completion Anonymous Call Anonymous
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Session Timer Call Hold Call Forward Call Transfer Network Conference Transfer on Conference Hang Up Directed Call Pickup Group Call Pickup Dialog Info Call Pickup Call Return Call Park Web Server Type Calling Line Identification Presentation Connected Line Identification Presentation DTMF Suppress DTMF Display Transfer via DTMF Intercom Power indicator LED indicates power statu
Configuring Basic Features this option is disabled, the status of the power indicator LED is determined by the option “Common Power Light On”. Hold/Held Power Light Flash Hold/Held Power Light Flash allows the power indicator LED to flash when a call is placed on hold or is held. If this option is disabled, the status of the power indicator LED is determined by the option “Common Power Light On”.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 7. Select the desired value from the pull-down list of Talk/Dial Power Light On. 8. Click Confirm to accept the change. Contrast determines the readability of the texts displayed on the LCD screen. Adjusting the contrast to a comfortable level can optimize the screen viewing experience. When configured properly, contrast allows users to read the LCD’s display with minimal eyestrain.
Configuring Basic Features 2. Select the desired value from the pull-down list of Contrast. 3. Click Confirm to accept the change. To configure contrast via phone user interface (applicable to SIP-T28P IP phones and EXP39 connected to SIP-T26P and SIP-T28P IP phones): 1. Press Menu->Settings->Basic Settings->Display->Contrast. 2. Press or , or the Switch soft key to increase or decrease the intensity of contrast. The default contrast level is 6. 3.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones enough time to read messages. Backlight active level is used to adjust the backlight intensity of the LCD screen. Backlight active level is only applicable to SIP-T28P IP phones and the connected EXP39. You can configure the backlight time as one of the following types: Always Off: Backlight is turned off permanently. Always On: Backlight is turned on permanently.
Configuring Basic Features To configure backlight via web user interface: 1. Click on Settings->Preference. 2. Select the desired value from the pull-down list of Backlight Active Level (only applicable to SIP-T28P IP phones and the connected EXP39). 3. Select the desired value from the pull-down list of Backlight Time (seconds). 4. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Procedure User password can be changed using the configuration files or locally. Change the user password of the Configuration File .cfg IP phone. For more information, refer to User Password on page 278. Change the user password of the IP phone. Local Web User Interface Navigate to: http:///servlet ?p=security&q=load To change the user password via web user interface: 1. Click on Security->Password. 2.
Configuring Basic Features Procedure Administrator password can be changed using the configuration files or locally. Change the administrator password. Configuration File .cfg For more information, refer to Administrator Password on page 278. Change the administrator password. Web User Interface Navigate to: http:///servlet Local ?p=security&q=load Phone User Interface Change the administrator password. To change the administrator password via web user interface: 1.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Phone lock is used to lock the IP phone to prevent it from unauthorized use. Once the IP phone is locked, a user must enter the password to unlock it. IP phones offer three types of phone lock: Menu Key, Function Keys and All Keys. The IP phone will not be locked immediately after the phone lock type is configured. One of the following steps is also needed: - Long press the pound key when the IP phone is idle.
Configuring Basic Features 0 Configure the type of phone Phone User Interface lock. Assign a keypad lock key. To configure phone lock via web user interface: 1. Click on Features->Phone Lock. 2. Select the desired type from the pull-down list of Keypad Lock Type. 3. Enter the unlock PIN in the Phone Unlock PIN (0~15 Digit) field. 4. Enter the desired time in the Phone Lock Time Out (0~3600s) field. 5. Click Confirm to accept the change. To configure a keypad lock key via web user interface: 1.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. In the desired DSS key field, select Keypad Lock from the pull-down list of Type. 3. Click Confirm to accept the change. To configure the type of phone lock via phone user interface: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Phone Settings->Keypad Lock. 2. Press or , or the Switch soft key to select the desired type from the Keypad Lock field. 3. Press the Save soft key to accept the change.
Configuring Basic Features them. The time and date display can use one of several different formats. Time Zone A time zone is a region on Earth that has a uniform standard time. It is convenient for areas in close commercial or other communication to keep the same time. When configuring the IP phone to obtain the time and date from the NTP server, you must set the time zone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Configure the NTP server, time zone and DST. Configure the time and date manually. Configure the time and date formats. For more information, refer to Time and Date on page 281. Configure NTP by DHCP priority feature. Configure the NTP server, time zone and DST. Configure the time and date Web User Interface manually. Configure the time and date formats.
Configuring Basic Features 2. Select the desired value from the pull-down list of NTP By DHCP Priority. 3. Click Confirm to accept the change. To configure the NTP server, time zone and DST via web user interface: 1. Click on Settings->Time & Date. 2. Select Disabled from the pull-down list of Manual Time. 3. Select the desired time zone from the pull-down list of Time Zone. 4. Enter the domain names or IP addresses in the Primary Server and Secondary Server fields respectively. 5.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones If you select Enabled, do one of the following: - Mark the DST By Date radio box in the Fixed Type field. Enter the start time in the Start Date field. Enter the end time in the End Date field. - Mark the DST By Week radio box in the Fixed Type field.
Configuring Basic Features 8. Click Confirm to accept the change. To configure the time and date manually via web user interface: 1. Click on Settings->Time & Date. 2. Select Enabled from the pull-down list of Manual Time. 3. Enter the time and date in the corresponding fields. 4. Click Confirm to accept the change. To configure the time and date format via web user interface: 1. Click on Settings->Time & Date. 2. Select the desired value from the pull-down list of Time Format. 3.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure the NTP server and time zone via phone user interface: 1. Press Menu->Settings->Basic Settings->Time & Date->SNTP Settings. 2. Press or , or the Switch soft key to select the time zone that applies to your area from the Time Zone field. The default time zone is "+8 China(Beijing)". 3. Enter the domain names or IP addresses in the NTP Server1 and NTP Server2 fields respectively. 4. Press the Save soft key to accept the change.
Configuring Basic Features Phone User Interface Web User Interface Portuguese SIP-T19P and SIP-T21P IP phones) Polish Spanish (not applicable to Spanish SIP-T19P and SIP-T21P IP phones) Turkish Turkish Not all of supported languages are available for selection. Languages available for selection depend on language packs currently loaded to the IP phone. You can make languages available for use on the phone user interface by loading language packs to the IP phone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The default language used on the phone user interface is English. The default language used on the web user interface depends on the language preferences in the browser (if the language is not supported by the IP phone, the web user interface uses English). You can specify the languages for the phone user interface and web user interface respectively.
Configuring Basic Features To specify the language for the phone user interface via phone user interface: 1. Press Menu->Settings->Basic Settings->Language. 2. Press 3. Press the Save soft key to accept the change. or to select the desired language. Logo customization allows unifying the IP phone appearance or displaying a custom image on the idle screen such as a company logo, instead of the default system logo. SIP-T20P IP phones only support a text logo.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure an image logo via web user interface (not applicable to SIP-T20P IP phones): 1. Click on Features->General Information. 2. Select Custom logo from the pull-down list of Use Logo. 3. Click Browse to select the logo file from your local system. 4. Click Upload to upload the file. 5. Click Confirm to accept the change. For SIP-T28P IP phones, the image logo is displayed on the idle screen.
Configuring Basic Features 3. Enter the desired text (0~15 characters) in the Text Logo field. 4. Click Confirm to accept the change. The registered account and the configured text logo are displayed alternately. Softkey layout is used to customize the soft keys at the bottom of the LCD screen to best meet users’ requirements. It can be configured based on call states. In addition to specifying which soft keys to display, you can determine their display order.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Call State Connecting Default Soft Keys Optional Soft Keys Empty Empty Empty Switch Empty Cancel Connecting SemiAttendTrans Transfer Empty Empty Switch Empty Cancel Dialing Send Empty IME History Delete Switch Cancel Line Favorite GPickup DPickup RingBack Empty Empty Empty Switch Empty CC Cancel RingBack SemiAttendTransBack Transfer Empty Empty Switch Empty CC Cancel Talk Transfer Empty Hold Mute Confer
Configuring Basic Features Call State Default Soft Keys Held Optional Soft Keys Empty Empty Empty Switch Empty Answer Cancel Reject NewCall PreTrans Conferenced Transfer Empty IME Directory Delete Switch Cancel Send Empty Empty Hold Switch 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. Configuration File .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones key and then click 7. . To adjust the display order of soft keys, select the desired soft key and then click or . The LCD screen displays the soft keys in the adjusted order. 8. Click Confirm to accept the change. Key as send allows assigning the pound key or star key as a send key. Send sound allows the IP phone to play a key tone when a user presses the send key. Key tone allows the IP phone to play a key tone when a user presses any key.
Configuring Basic Features http:///servlet ?p=features-audio&q=load Phone User Interface Configure the send key. To configure a send key via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Key As Send. 3. Click Confirm to accept the change. To configure a send sound and key tone via web user interface: 1. Click on Features->Audio. 2. Select the desired value from the pull-down list of Key Sound.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 3. Select the desired value from the pull-down list of Send Sound. 4. Click Confirm to accept the change. To configure send key via phone user interface: 1. Press Menu->Features->Key as Send. 2. Press or , or the Switch soft key to select # or * from the Key as Send field, or select Disable to disable this feature. 3. Press the Save soft key to accept the change.
Configuring Basic Features IP phone waits before automatically dial out the hotline number. Navigate to: http:///servlet ?p=features-general&q=load Configure the hotline number. Specify the time (in seconds) the Phone User Interface IP phone waits before automatically dialing out the hotline number. To configure hotline via web user interface: 1. Click on Features->General Information. 2. Enter the hotline number in the Hotline Number field. 3.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Call log contains call information such as remote party identification, time and date, and call duration. IP phones maintain a local call log. Call log consists of four lists: Placed Calls, Received Calls, Missed Calls and Forwarded Calls. Call log lists support 100 entries in all. To store call information, you must enable save call log feature in advance. Procedure Call log can be configured using the configuration files or locally.
Configuring Basic Features To configure call log feature via phone user interface: 1. Press Menu->Features->History Setting. 2. Press or , or the Switch soft key to select the desired value from the History Record field. 3. Press the Save soft key to accept the change. Missed call log allows the IP phone to display the number of missed calls with an indicator icon on the idle screen, and to log missed calls in the Missed Calls list when the IP phone misses calls.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 4. Select the desired value from the pull-down list of Missed Call Log. 5. Click Confirm to accept the change. IP phones maintain a local directory. The local directory can store up to 1000 contacts and 5 groups. When adding a contact to the local directory, in addition to name and phone numbers, you can also specify the account, ring tone and group for the contact.
Configuring Basic Features To add a group to the local directory via web user interface: 1. Click on Directory->Local Directory. 2. In the Group Setting block, enter the desired group name in the Group field. 3. Select the desired ring tone from the pull-down list of Ring field. 4. Click Add to add the group. To add a contact to the local directory via web user interface: 1. Click on Directory->Local Directory. 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 5. Select the desired account from the pull-down list of Account. If Auto is selected, the IP phone will use the first available account when placing calls to the contact from the local directory. 6. Click Add to add the contact. To add a group to the local directory via phone user interface: 1. Press Menu->Directory->Local Directory. 2. Press the AddGrp soft key. 3. Enter the desired group name in the Name field. 4.
Configuring Basic Features Live dialpad allows IP phones to automatically dial out the entered phone number after a specified period of time. Procedure Live dialpad can be configured using the configuration files or locally. Configure live dialpad. Configuration File .cfg For more information, refer to Live Dialpad on page 294. Configure live dialpad.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones enabled. Procedure Call waiting and call waiting tone can be configured using the configuration files or locally. Configure call waiting and call Configuration File .cfg waiting tone. For more information, refer to Call Waiting on page 295. Configure call waiting. Navigate to: http:///servlet Web User Interface Local ?p=features-general&q=load Configure call waiting tone.
Configuring Basic Features 4. (Optional.) Enter the call waiting off code in the Call Waiting Off Code field. 5. Click Confirm to accept the change. To configure call waiting tone via web user interface: 1. Click on Features->Audio. 2. Select the desired value from the pull-down list of Call Waiting Tone. 3. Click Confirm to accept the change. To configure call waiting and call waiting tone via phone user interface: 1. Press Menu->Features->Call Waiting. 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Tone field. 4. (Optional.) Enter the call waiting on code in the CW On Code field. 5. (Optional.) Enter the call waiting off code in the CW Off Code field. 6. Press the Save soft key to accept the change. Auto redial allows IP phones to redial a busy number after the first attempt. Both the number of attempts and waiting time between redials are configurable. Procedure Auto redial can be configured using the configuration files or locally.
Configuring Basic Features 4. Enter the desired times in the Auto Redial Times (1~300) field. The default value is 10. 5. Click Confirm to accept the change. To configure auto redial via phone user interface: 1. Press Menu->Features->Auto Redial. 2. Press or , or the Switch soft key to select the desired value from the Auto Redial field. 3. Enter the waiting time (in seconds) in the Redial Interval field. 4. Enter the desired times in the Redial Times field. 5.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Specify a period of delay time for .cfg auto answer. For more information, refer to Auto Answer on page 297. Configure auto answer. Navigate to: http:///servlet ?p=account-basic&q=load&acc Local Web User Interface =0 Specify a period of delay time for auto answer. Navigate to: http://servlet? p=features-general&q=load Phone User Interface Configure auto answer.
Configuring Basic Features 2. Enter the desired time in the Auto-Answer Delay (1~4s) field. 3. Click Confirm to accept the change. To configure auto answer via phone user interface: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Accounts. 2. Select the desired account and then press the Enter soft key. 3. Press or , or the Switch soft key to select the desired value from the Auto Answer field. 4. Press the Save soft key to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Procedure Call completion can be configured using the configuration files or locally. Configure call completion. Configuration File .cfg For more information, refer to Call Completion on page 298. Configure call completion. Web User Interface Local Navigate to: http:///servlet ?p=features-general&q=load Phone User Interface Configure call completion. To configure call completion via web user interface: 1.
Configuring Basic Features Anonymous call allows the caller to conceal the identity information displayed on the callee’s screen. The callee’s phone LCD screen prompts an incoming call from anonymity. Anonymous call is configurable on a per-line basis. Example of anonymous SIP header: Via: SIP/2.0/UDP 10.2.8.183:5063;branch=z9hG4bK1535948896 From: "Anonymous" ;tag=128043702 To: Call-ID: 1773251036@10.2.8.183 CSeq: 1 INVITE Contact:
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure anonymous call via web user interface: 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3. Click on Basic. 4. Select the desired value from the pull-down list of Local Anonymous. 5. Select the desired value from the pull-down list of Send Anonymous Code. 6. (Optional.) Enter the anonymous call on code in the On Code field. 7. (Optional.
Configuring Basic Features The anonymous call rejection on code and anonymous call rejection off code configured on IP phones are used to activate/deactivate the server-side anonymous call rejection feature. They may vary on different servers. Procedure Anonymous call rejection can be configured using the configuration files or locally. Configure anonymous call rejection. Configuration File .cfg For more information, refer to Anonymous Call Rejection on page 301. Configure anonymous call rejection.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure anonymous call rejection via phone user interface: 1. Press Menu->Features->Anonymous Call. 2. Press or , or the Switch soft key to select the desired line from the Line ID or , or the Switch soft key to select the desired value from the Anon field. 3. Press Rejection field. 4. (Optional.) Enter the anonymous call rejection on code in the Reject On Code field. 5. (Optional.
Configuring Basic Features Configure the DND mode. Configure DND in the phone mode. Specify the return code and the reason of the SIP response message when DND is enabled. For more information, refer to Do Not Disturb on page 302. Assign a DND key. Navigate to: http:///servlet? p=dsskey&q=load&model=0 Configure DND.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. In the desired DSS key field, select DND from the pull-down list of Type. 3. Click Confirm to accept the change. To configure DND feature via web user interface: 1. Click on Features->Forward & DND. 2. In the DND block, mark the desired radio box in the Mode field. a) If you mark the Phone radio box: 1) Mark the desired radio box in the DND Status field. 2) (Optional.) Enter the DND on code in the DND On Code field. 3) (Optional.
Configuring Basic Features 2) Mark the desired radio box in the DND Status field. 3) (Optional.) Enter the DND on code in the DND On Code field. 4) (Optional.) Enter the DND off code in the DND Off Code field. 3. Click Confirm to accept the change. To specify the return code and the reason when DND is enabled via web user interface: 1. Click on Features->General Information. 2. Select the desired type from the pull-down list of Return Code When DND. 3. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure a DND key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2. Select the desired DSS key. 3. Press or , or the Switch soft key to select Key Event from the Type field. 4. Press or , or the Switch soft key to select DND from the Key Type field. 5. Press the Save soft key to accept the change. To configure DND in the phone mode via phone user interface: 1.
Configuring Basic Features 2． Select the desired value from the pull-down list of Busy Tone Delay (Seconds). 3． Click Confirm to accept the change. Return code when refuse defines the return code and reason of the SIP response message for the refused call. The caller’s phone LCD screen displays the reason according to the received return code.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones http:///servlet ?p=features-general&q=load To specify the return code and the reason when refusing a call via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Return Code When Refuse. 3. Click Confirm to accept the change. Early media refers to media (e.g., audio and video) played to the caller before a SIP call is actually established.
Configuring Basic Features Procedure 180 ring workaround can be configured using the configuration files or locally. Configure 180 ring workaround. Configuration File .cfg For more information, refer to 180 Ring Workaround on page 306. Configur 180 ring workaround. Local Web User Interface Navigate to: http:///servlet ?p=features-general&q=load To configure 180 ring workaround via web user interface: 1. Click on Features->General Information. 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones An outbound proxy server can receive all initiating request messages and route them to the designated destination. If the IP phone is configured to use an outbound proxy server within a dialog, all SIP request messages from the IP phone will be sent to the outbound proxy server forcefully. Note To use this feature, make sure the outbound server has been correctly configured on the IP phone.
Configuring Basic Features 2. Select the desired value from the pull-down list of Use Outbound Proxy In Dialog. 3. Click Confirm to accept the change. SIP session timers T1, T2 and T4 are SIP transaction layer timers defined in RFC 3261. Timer T1 is an estimate of the Round Trip Time (RTT) of transactions between a SIP client and SIP server. Timer T2 represents the maximum retransmit interval for non-INVITE requests and INVITE responses.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure session timer via web user interface: 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3. Click on Advanced. 4. Enter the desired value in the SIP Session Timer T1 (0.5~10s) field. The default value is 0.5s. 5. Enter the desired value in the SIP Session Timer T2 (2~40s) field. The default value is 4s. 6. Enter the desired value in the SIP Session Timer T4 (2.5~60s) field.
Configuring Basic Features Procedure Session timer can be configured using the configuration files or locally. Configure session timer. Configuration File .cfg For more information, refer to Session Timer on page 308. Configure session timer. Navigate to: Local Web User Interface http:///servlet ?p=account-adv&q=load&acc= 0 To configure session timer via web user interface: 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Call hold provides a service of placing an active call on hold. When a call is placed on hold, the IP phone sends an INVITE request with a HOLD SDP to the server. IP phones support two call hold methods, one is RFC 3264, which sets the “a” (media attribute) in the SDP to sendonly, recvonly or inactive (e.g., a=sendonly). The other is RFC 2543, which sets the “c” (connection addresses for the media streams) in the SDP to zero (e.g., c=0.0.0.0).
Configuring Basic Features 2. Select the desired value from the pull-down list of RFC 2543 Hold. 3. Click Confirm to accept the change. To configure call hold tone and call hold tone delay via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Play Hold Tone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 3. Enter the desired time in the Play Hold Tone Delay field. 4. Click Confirm to accept the change. Call forward allows users to redirect an incoming call to a third party. IP phones redirect an incoming INVITE message by responding with a 302 Moved Temporarily message, which contains a Contact header with a new URI that should be tried. Three types of call forward: Always Forward -- Forward the incoming call immediately.
Configuring Basic Features Procedure Call forward can be configured using the configuration files or locally. Configure call forward in .cfg custom mode. For more information, refer to Call Forward on page 311. Configure the call forward mode. Configuration File Configure call forward in .cfg phone mode. Configure forward international. For more information, refer to Call Forward on page 311. Configure call forward.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 4) Select the ring time to wait before forwarding from the pull-down list of After Ring Time (0~120s) (only for the no answer forward). b) If you mark the Custom radio box: 1) Select the desired account from the pull-down list of Account. 2) Mark the desired radio box in the Always/Busy/No Answer Forward field. 3) Enter the destination number you want to forward in the Target field.
Configuring Basic Features 2. Select the desired value from the pull-down list of Fwd International. 3. Click Confirm to accept the change. To configure call forward in phone mode via phone user interface: 1. Press Menu->Features->Call Forward. 2. Press or to select the desired forwarding type, and then press the Enter soft key. 3. Depending on your selection: a) If you select Always Forward: 1) Press or , or the Switch soft key to select the desired value from the Always field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones No Answer field. 2) Enter the destination number you want to forward all unanswered incoming calls to in the Forward To field. 3) Press or , or the Switch soft key to select the ring time to wait before forwarding from the After Ring Time field. The default ring time is 12 seconds. 4) (Optional.) Enter the no answer forward on code and off code respectively in the On Code and Off Code fields. 4. Press the Save soft key to accept the change.
Configuring Basic Features You can also configure the busy forward for all accounts. After the busy forward was configured for a specific account, do the following: 1) Press or to highlight the Busy field. 2) Press the All Lines soft key. The LCD screen prompts “Copy to All Lines?”. 3) Press the OK soft key to accept the change. c) If you select No Answer Forward, you can configure it for a specific account. 1) Press or , or the Switch soft key to select the desired value from the No Answer field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones transfer through on-hook. When a user performs a semi-attended transfer, semi-attended transfer feature determines whether to display the prompt "n New Missed Call(s)" ("n" indicates the number of the missed calls) on the destination party’s phone LCD screen. Procedure Call transfer can be configured using the configuration files or locally. Specify whether to complete the transfer through on-hook. Configuration File .
Configuring Basic Features Network conference, also known as centralized conference, provides users with flexibility of call with multiple participants (more than three). IP phones implement network conference using the REFER method specified in RFC 4579. This feature depends on support from a SIP server. Procedure Network conference can be configured using the configuration files or locally. Configure network conference. Configuration File .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 5. Enter the conference URI in the Conference URI field. 6. Click Confirm to accept the change. For local conference, all parties drop the call when the conference initiator drops the conference call. Transfer on conference hang up allows the other two parties to remain connected when the conference initiator drops the conference call. Procedure Transfer on conference hang up can be configured using the configuration files or locally.
Configuring Basic Features To configure Transfer on Conference Hang up via web user interface: 1. Click on Features->Transfer. 2. Select the desired value from the pull-down list of Transfer on Conference Hang up. 3. Click Confirm to accept the change. Directed call pickup is used for picking up an incoming call on a specific extension. A user can pick up the incoming call using a directed pickup key or the DPickup soft key (not applicable to SIP-T20P IP phones).
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Assign a directed call pickup key. .cfg For more information, refer to Directed Call Pickup Key on page 409. Assign a directed call pickup key. Navigate to: http:///servl et?p=dsskey&q=load&model= 0 Configure directed call pickup feature on a phone basis. Web User Interface Local Navigate to: http:///servl et?p=features-callpickup&q=lo ad Configure directed call pickup code on a per-line basis.
Configuring Basic Features 4. Select the desired line from the pull-down list of Line. 5. Click Confirm to accept the change. To configure directed call pickup feature on a phone basis via web user interface: 1. Click on Features->Call Pickup. 2. Select the desired value from the pull-down list of Directed Call Pickup (not applicable to SIP-T20P IP phones). 3. Enter the directed call pickup code in the Directed Call Pickup Code field. 4. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 4. Enter the directed call pickup code in the Directed Call Pickup Code field. 5. Click Confirm to accept the change. To configure a directed pickup key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2. Select the desired DSS key. 3. Press or , or the Switch soft key to select Key Event from the Type field. 4.
Configuring Basic Features Procedure Group call pickup can be configured using the configuration files or locally. Configure the group call pickup code on a per-line basis. .cfg Configure group call pickup feature on a phone basis. For more information, refer to Configuration File Group Call Pickup on page 324. Assign a group call pickup key. .cfg For more information, refer to Group Call Pickup Key on page 411. Assign a group call pickup key.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 4. Select the desired line from the pull-down list of Line. 5. Click Confirm to accept the change. To configure group call pickup feature on a phone basis via web user interface: 1. Click on Features->Call Pickup. 2. Select the desired value from the pull-down list of Group Call Pickup (not applicable to SIP-T20P IP phones). 3. Enter the group call pickup code in the Group Call Pickup Code field. 4. Click Confirm to accept the change.
Configuring Basic Features 4. Enter the group call pickup code in the Group Call Pickup Code field. 5. Click Confirm to accept the change. To configure a group pickup key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2. Select the desired DSS key. 3. Press or , or the Switch soft key to select Key Event from the Type field. 4.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Example of the dialog-info message carried in NOTIFY message:
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. In the desired DSS key field, select Call Return from the pull-down list of Type. 3. Click Confirm to accept the change. To configure a call return key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2. Select the desired DSS key. 3. Press or , or the Switch soft key to select Key Event from the Type field. 4. Press or , or the Switch soft key to select Call Return from the Key Type field.
Configuring Basic Features Assign a call park key. Navigate to: Web User Interface Local http:///servl et?p=dsskey&q=load&model= 0 Phone User Interface Assign a call park key. To configure a call park key via web user interface: 1. Click on DSSKey->Memory Key (or Line Key). 2. In the desired DSS key field, select Call Park from the pull-down list of Type. 3. Enter the desired value (e.g., call park feature code) in the Value field. 4.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 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 pages returned by the web server. Both HTTP and HTTPS port numbers are configurable. Procedure Web server type can be configured using the configuration files or locally. Configure the web access type, Configuration File .cfg HTTP port and HTTPS port.
Configuring Basic Features 5. Enter the HTTPS port number in the HTTPS Port (1~65535) field. The default HTTPS port number is 443. 6. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. 7. Click OK to reboot the IP phone. To configure web server type via phone user interface: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->Webserver Type. 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones identity in the relevant SIP header. If the caller has existed in the local directory, the local name assigned to the caller should be preferentially displayed. Procedure CLIP can be configured using the configuration files or locally. Configure the presentation of the caller identity. Configuration File .cfg For more information, refer to Calling Line Identification Presentation on page 327.
Configuring Basic Features 5. Click Confirm to accept the change. Connected line identification presentation (COLP) allows IP phones to display the identity of the callee specified for outgoing calls. IP phones 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.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Three 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. SIP INFO -- DTMF digits are transmitted by SIP INFO messages. The method of transmitting DTMF digits is configurable on a per-line basis. RFC 2833 DTMF digits are transmitted using the RTP Event packets that are sent along with the voice path.
Configuring Basic Features Configure the method of transmitting DTMF digits and the payload type. Navigate to: http:///servl et?p=account-adv&q=load&ac Local Web User Interface c=0 Configure the number of times for the IP phone to send the end RTP Event packet. Navigate to: http:///servl et?p=features-general&q=loa d To configure the method of transmitting DTMF digits via web user interface: 1. Click on Account. 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 7. Click Confirm to accept the change. To configure the number of times to send the end RTP Event packet via web user interface: 1. Click on Features->General Information. 2. Select the desired value (1-3) from the pull-down list of DTMF Repetition. 3. Click Confirm to accept the change. Suppress DTMF display allows IP phones to suppress the display of DTMF digits. DTMF digits are displayed as “*” on the LCD screen.
Configuring Basic Features display delay. Navigate to: http:///servl et?p=features-general&q=loa d To configure suppress DTMF display and suppress DTMF display delay via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Suppress DTMF Display. 3. Select the desired value from the pull-down list of Suppress DTMF Display Delay (not applicable to SIP-T20P IP phones). 4. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Configure transfer via DTMF. Navigate to: Local Web User Interface http:///servl et?p=features-general&q=loa d To configure transfer via DTMF via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of DTMF Replace Tran. 3. Enter the specified DTMF digits in the Tran Send DTMF field. 4. Click Confirm to accept the change.
Configuring Basic Features Procedure Intercom key can be configured using the configuration files or locally. Assign an intercom key. Configuration File .cfg For more information, refer to Intercom Key on page 415. Assign an intercom key. Web User Interface Local Navigate to: http:///servlet ?p=dsskey&q=load&model=0 Phone User Interface Assign an intercom key. To configure an intercom key via web user interface: 1. Click on DSSKey->Memory Key (or Line Key). 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The IP phone can process incoming calls differently depending on settings. There are four configuration options for incoming intercom calls: 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. Intercom Tone Intercom Tone allows the IP phone to play a warning tone before answering an intercom call.
Configuring Basic Features 2. Select the desired values from the pull-down lists of Accept Intercom, Intercom Mute, Intercom Tone and Intercom Barge. 3. Click Confirm to accept the change. To configure intercom via phone user interface: 1. Press Menu->Features->Intercom. 2. Press or , or the Switch soft key to select the desired values from the Accept Intercom, Intercom Mute, Intercom Tone and Intercom Barge fields. 3. Press the Save soft key to accept the change.
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Configuring Advanced Features This chapter provides information for making configuration changes for the following advanced features: Distinctive Ring Tones Tones Remote Phone Book LDAP Busy Lamp Field Music on Hold Automatic Call Distribution Message Waiting Indicator Multicast Paging Call Recording Hot Desking Action URL Action URI Server Redundancy LLDP VLAN VPN Quality of Service Network Address Translation 802.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Alert-Info headers in the following two formats: Alert-Info: http://localIP/Bellcore-drN Alert-Info: ;info=info text;x-line-id=0 If the Alter-Info header contains the keyword “Bellcore-drN”, the IP phone will play the Bellcore-drN ring tone (N=1, 2, 3, 4 or 5) (if the parameter “features.alert_info_tone” is set to 1). Example: Alert-Info: http://127.0.0.
Configuring Advanced Features If the Alert-Info header contains a remote URL, the IP phone will try to download the WAV ring tone file from the URL and then play the remote ring tone (if the parameter “account.X.alert_info_url_enable” is set to 1). If it fails to download the file, the IP phone will play the local ring tone associated with info text. If there is no text matched, the IP phone will play the preconfigured local ring tone in about ten seconds. Example: Alert-Info: http:
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 4. Select the desired value from the pull-down list of Distinctive Ring Tones. 5. Click Confirm to accept the change. To configure the internal ringer text and internal ringer file via web user interface: 1. Click on Settings->Ring. 2. Enter the keywords in the Internal Ringer Text fields. 3. Select the desired ring tones for each text from the pull-down lists of Internal Ringer File.
Configuring Advanced Features 4. Click Confirm to accept the change. When receiving a message, the IP phone will play a warning tone. You can customize tones or select specialized tone sets (vary from country to country) to indicate different conditions of the IP phone. The default tones used on IP phones are the US tone sets.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Chile Czech ETSI Configured tones can be heard on IP phones for the following conditions.
Configuring Advanced Features 2. Select the desired type from the pull-down list of Select Country. If you select Custom, you can customize a tone for each condition of the IP phone. 3. Click Confirm to accept the change. Remote phone book is a centrally maintained phone book, stored on the remote server. Users only need the access URL of the remote phone book.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones receives an incoming call. Specify how often the IP phone refreshes the local cache of the remote phone book. For more information, refer to Remote Phone Book on page 338. Specify the access URL of the remote phone book. Navigate to: http:///servl et?p=contacts-remote&q=load Specify whether to query the entry name from the remote Local Web User Interface phone book when the IP phone receives an incoming call.
Configuring Advanced Features To configure Search Remote Phonebook Name and Search Flash Time via web user interface: 1. Click on Directory->Remote Phone Book. 2. Select the desired value from the pull-down list of Search Remote Phonebook Name. 3. Enter the desired time in the Search Flash Time (Seconds) field. 4. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones LDAP Attributes The following table lists the most common attributes used to configure the LDAP lookup on IP phones. Abbreviation Name Description gn givenName First name cn commonName sn surname dn distinguishedName dc dc - company - telephoneNumber mobile mobilephoneNumber ipPhone IPphoneNumber LDAP attribute is made up from given name joined to surname.
Configuring Advanced Features 2. Enter the values in the corresponding fields. 3. Select the desired values from the corresponding pull-down lists. 4. Click Confirm to accept the change. To configure an LDAP key via web user interface: 1. Click on DSSKey->Memory Key (Line Keys or Programable Key). SIP-T19P IP phones only support programable keys and SIP-T22P/T21P/T20P IP phones only support line keys and programable keys. 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 3. Press or , or the Switch soft key to select Key Event from the Type field. 4. Press or , or the Switch soft key to select LDAP from the Key Type field. 5. Press the Save soft key to accept the change. Busy Lamp Field (BLF) is used to monitor a specific user for status changes on IP phones. For example, you can configure a BLF key on a supervisor’s phone to monitor the phone user status (busy or idle).
Configuring Advanced Features LED Status Description The monitored user’s conversation is placed on hold. Slow flashing green (1s) Off The call is parked against the monitored user’s phone number. The monitored user does not exist. Memory key LED (configured as a BLF key and BLF LED Mode is set to 0) LED Status Description Solid green The monitored user is idle. Fast flashing red (200ms) The monitored user receives an incoming call. Solid red The monitored user is dialing.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones LED Status Off Description The monitored user is idle. The monitored user does not exist. Line key LED (configured as a BLF key and BLF LED Mode is set to 2) LED Status Description Fast flashing green (200ms) The monitored user receives an incoming call. The monitored user is dialing. Slow flashing green (500ms) The monitored user is talking. The monitored user’s conversation is placed on hold.
Configuring Advanced Features Memory key LED (configured as a BLF key and BLF LED Mode is set to 3) LED Status Description Fast flashing red (200ms) The monitored user receives an incoming call. The monitored user is dialing. Solid red The monitored user is talking. The monitored user’s conversation is placed on hold. Slow flashing red (1s) Off The call is parked against the monitored user’s phone number. The monitored user is idle. The monitored user does not exist.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones http:///servl et?p=features-general&q=loa d Phone User Interface Assign a BLF key. To configure a BLF key via web user interface: 1. Click on DSSKey->Memory Key (or Line Key). 2. In the desired DSS key field, select BLF from the pull-down list of Type. 3. Enter the phone number or extension you want to monitor in the Value field. 4. Select the desired line from the pull-down list of Line. 5. (Optional.
Configuring Advanced Features 3. Select the desired value from the pull-down list of Audio Alert for BLF Pickup. 4. Click Confirm to accept the change. To configure BLF LED mode via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of BLF LED Mode. 3. Click Confirm to accept the change. To configure a BLF key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2. Select the desired DSS key.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 3. Press or , or the Switch soft key to select BLF from the Type field. 4. Press or , or the Switch soft key to select the desired line from the Account ID field. 5. Enter the phone number or extension you want to monitor in the Value field. 6. (Optional.) Enter the directed call pickup code in the Extension field. 7. Press the Save soft key to accept the change.
Configuring Advanced Features 4. Enter the SIP URI (e.g., sip:moh@sip.com) in the Music Server URI field. 5. Click Confirm to accept the change. Automatic Call Distribution (ACD) enables organizations to manage a large number of phone calls on an individual basis. ACD enables the use of IP phones in a call-center role by automatically distributing incoming calls to available users, or agents. ACD depends on support from a SIP server. ACD is disabled on the phone by default.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Procedure ACD can be configured using the configuration files or locally. Assign an ACD key. For more information, refer to Configuration File .cfg ACD Key on page 419. Configure ACD auto available. For more information, refer to ACD on page 348. Assign an ACD key. Navigate to: http:///servlet Local Web User Interface ?p=dsskey&q=load&model=0 Configure ACD auto available.
Configuring Advanced Features 3. Enter the desired time in ACD Auto Available Timer (0~120s) field. 4. Click Confirm to accept the change. To configure an ACD key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2. Select the desired DSS key. 3. Press 4. Press the Save soft key to accept the change. or , or the Switch soft key to select ACD from the Type field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Procedure Configuration changes can be performed using the configuration files or locally. Configure subscribe for MWI. Configure subscribe MWI to voice Configuration File .cfg mail. For more information, refer to Message Waiting Indicator on page 348. Configure subscribe for MWI. Configure subscribe MWI to voice mail.
Configuring Advanced Features 6. Click Confirm to accept the change. To configure subscribe MWI to voice mail via web user interface: 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3. Click on Advanced. 4. Select the desired value from the pull-down list of Subscribe MWI To Voice Mail. 5. Enter the desired voice number in the Voice Mail field. 6. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Procedure Configuration changes can be performed using the configuration files or locally. Assign a multicast paging key. For more information, refer to Multicast Paging Key on page Configuration File .cfg 419. Specify a multicast codec for the IP phone to use for multicast RTP. For more information, refer to Sending RTP Stream on page 351. Assign a multicast paging key.
Configuring Advanced Features To configure a codec for multicast paging via web user interface: 1. Click on Features->General Information. 2. Select the desired codec from the pull-down list of Multicast Codec. 3. Click Confirm to accept the change. To configure a multicast paging key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2. Select the desired DSS key. 3. Press or , or the Switch soft key to select Key Event from the Type field. 4.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones will be automatically ignored. If the parameter is the priority value, the incoming multicast paging calls with higher priority are automatically answered and the ones with lower priority are ignored. Paging Priority Active This parameter decides how the IP phone handles the incoming multicast paging calls when there is already a multicast paging call in progress.
Configuring Advanced Features 3. Enter the label in the Label field. The label will appear on the LCD screen when receiving the RTP multicast. 4. Click Confirm to accept the change. To configure paging barge and paging priority active features via web user interface: 1. Click on Directory->Multicast IP. 2. Select the desired value from the pull-down list of Paging Barge. 3. Select the desired value from the pull-down list of Paging Priority Active. 4. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Call recording enables users to record calls. It depends on support from a SIP server. When the user presses the call record key, the IP phone sends a record request to the server. IP phones themselves do not have memory to store the recording, what they can do is to trigger the recording and indicate the recording status. Normally, there are 2 main methods to trigger a recording on a certain server. We call them record and URL record.
Configuring Advanced Features User-Agent: Yealink SIP-T28P 2.72.0.1 Record: off Content-Length: 0 URL Record When a user presses a URL record key for the first time during a call, the IP phone sends an HTTP GET message to the server. Example of an HTTP GET message: Get /phonerecording.cgi?model=yealink HTTP/1.0\r\n Request Method: GET Request URI: /phonerecording.cgi?model=yealink Request version: HTTP/1.0 Host: 10.1.2.224\r\n User-agent: yealink SIP-T28P 2.72.0.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 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 a 200 OK message. Example of a 200 OK message: The recording session is successfully stopped. Procedure Call recording key can be configured using the configuration files or locally. Assign a record key.
Configuring Advanced Features 2. In the desired DSS key field, select Record from the pull-down list of Type. 3. Click Confirm to accept the change. To configure a URL record key via web user interface: 1. Click on DSSKey->Memory Key (or Line Key). 2. In the desired DSS key field, select URL Record from the pull-down list of Type. 3. Enter the URL in the Value field. 4. Click Confirm to accept the change. To configure a record key via phone user interface: 1.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure a URL record key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2. Select the desired DSS key. 3. Press 4. Enter the URL in the Value field. 5. Press the Save soft key to accept the change. or , or the Switch soft key to select URL Record from the Type field.
Configuring Advanced Features 2. In the desired DSS key field, select Hot Desking from the pull-down list of Type. 3. Click Confirm to accept the change. Note You can configure a programable key as a hot desking key on SIP-T19P IP phones only. To configure a hot desking key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2. Select the desired DSS key. 3. Press or , or the Switch soft key to select Key Event from the Type field. 4.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Event On Hook When the IP phone is on hook. Incoming Call When the IP phone receives an incoming call. Outgoing Call When the IP phone places a call. Established When the IP phone establishes a call. Terminated When the IP phone terminates a call. Open DND When the IP phone enables the DND mode. Close DND When the IP phone disables the DND mode. Open Always Forward When the IP phone enables the always forward.
Configuring Advanced Features An HTTP or HTTPS GET request may contain variable name and variable value, separated by “=”. Each variable value starts with $ in the query part of the URL. The valid URL format is: http(s)://IP address of server/help.xml?variable name=$variable. Variable name can be customized by users, while the variable value is pre-defined. For example, a URL “http://192.168.1.10/help.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Variable Value $call_id Description The call-id of the active call. Procedure Action URL can be configured using the configuration files or locally. Configure action URL. Configuration File .cfg For more information, refer to Action URL on page 354. Configure action URL. Navigate to: Local Web User Interface http:///servl et?p=features-actionurl&q=loa d To configure action URL via web user interface: 1.
Configuring Advanced Features which are separated by “=”. The valid URI format is: http(s)://phone IP address/servlet?key=variable value. The following table lists pre-defined variable values: Variable Value Phone Action OK Press the OK key (For SIP-T19P, press ENTER Press the Enter soft key (Except for SIP-T20P). SPEAKER Press the Speakerphone key. F_TRANSFER Transfers a call to another party. VOLUME_UP Increase the volume. VOLUME_DOWN Decrease the volume. MUTE Mute the call.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Variable Value Phone Action ONHOOK Hang up the handset. ANSWER Answer a call. Reset Reset a phone. Perform a semi-attended/attended transfer to ATrans=xxx xxx. BTrans=xxx Perform a blind transfer to xxx. CALLEND End a call. Get firmware version, registration, DND or forward configuration information. The valid value of “x” is 0 or 1, 0 means you do phonecfg=get[&accounts=x][&dnd =x][&fw=x] not need to get configuration information.
Configuring Advanced Features Specify the trusted IP address(es) for sending the action URI to the IP phone. Local Web User Interface Navigate to: http:///servl et?p=features-remotecontrl&q =load To configure the trusted IP address(es) for action URI via web user interface: 1. Click on Features->Remote Control. 2. Enter the IP address or any in the Action URI allow IP List field. Multiple IP addresses are separated by commas.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Fallback: In this mode, a second less featured call server with SIP capability takes over call control to provide basic calling capability, but without some advanced features offered by the working server (for example, shared line, call recording and MWI). IP phones support configuration of two SIP servers per SIP registration for fallback purpose.
Configuring Advanced Features Phone Registration Registration methods of the fallback mode include: Concurrent registration: The IP phone registers to two SIP servers (working server and fallback server) at the same time. In a failure situation, a fallback server can take over the basic calling capability, but without some of the advanced features offered by the working server (default registration method). Successive registration: The IP phone only registers to one server at a time.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 5. Configure parameters of SIP server 1 and SIP server 2 in the corresponding fields. 6. Click Confirm to accept the change. To configure server redundancy for failover purpose via web user interface: 1. Click on Account->Register. 2. Select the desired account from the pull-down list of Account. 3. Configure registration parameters of the selected account in the corresponding fields. 4.
Configuring Advanced Features 5. Configure parameters of the SIP server 1 or SIP server 2 in the corresponding fields. You must set the port of SIP server to 0 for NAPTR, SRV and A queries. 6. Note Click Confirm to accept the change. If the outbound proxy server is required and the transport is set to DNS-NAPTR, you must set the port of outbound proxy server to 0 for NAPTR, SRV and A queries.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones NAPTR (Naming Authority Pointer) First, the IP phone sends NAPTR query to get the NAPTR pointer and transport protocol. Example of NAPTR records: order pref flags service regexp replacement IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.yealink.pbx.com IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.yealink.pbx.
Configuring Advanced Features Parameters are explained in the following table: Parameter Priority Description Specify preferential treatment for the specific host entry. Lower priority is more preferred. When priorities are equal, weight is used to differentiate the Weight preference. The preference is from highest to lowest. Keep the same to load balance. Port Target Identify the port number to be used. Identify the actual host for an A query. SRV query returns two records.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones If it is not the last server in the list, the maximum number of retries depends on the configured retry count. Procedure Server redundancy can be configured using the configuration files or locally. Configure the transport type on the IP phone. Configuration File .cfg For more information, refer to SIP Server Domain Name Resolution on page 360. Configure the transport type on the IP phone.
Configuring Advanced Features their attributes such as model number, serial number and software revision. TLVs supported by IP phones are summarized in the following table: TLV Type Mandatory TLVs TLV Name Description Chassis ID The network address of the IP phone. Port ID The MAC address of the IP phone. Time To Live Seconds until data unit expires. End of LLDPDU Marks end of LLDPDU. System Name System Description Name assigned to the IP phone. The default value is “yealink”.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones TLV Type TLV Name Description and DSCP value. Extended Power-via-MDI Inventory – Hardware Revision Inventory – Firmware Revision Inventory – Software Revision Inventory – Serial Number Power type, source, priority and value. Hardware revision of the IP phone. Firmware revision of the IP phone. Software revision of the IP phone. Serial number of the IP phone. Inventory – Manufacturer name of the IP phone.
Configuring Advanced Features To configure LLDP via web user interface: 1. Click on Network->Advanced. 2. In the LLDP block, select the desired value from the pull-down list of Active. 3. Enter the desired time interval in the Packet Interval (1~3600s) field. 4. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. 5. Click OK to reboot the IP phone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones In addition to manual configuration, the IP phone also supports automatic discovery of VLAN via LLDP or DHCP. The assignment takes place in this order: assignment via LLDP, manual configuration, then assignment via DHCP. VLAN Discovery via DHCP IP phones support VLAN discovery via DHCP. When the VLAN Discovery method is set to DHCP, the IP phone will examine DHCP option for a valid VLAN ID.
Configuring Advanced Features 4. Select the desired value (0-7) from the pull-down list of Priority. 5. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after a reboot. 6. Click OK to reboot the IP phone. To configure VLAN for PC port via web user interface: 1. Click on Network->Advanced. 2. In the VLAN block, select the desired value from the pull-down list of PC Port Active. 3. Enter the VLAN ID in the VID (1-4094) field. 4.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To configure DHCP VLAN discovery via web user interface: 1. Click on Network->Advanced. 2. In the VLAN block, select the desired value from the pull-down list of DHCP VLAN Active. 3. Enter the desired option in the Option field. The default option is 132. 4. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. 5. Click OK to reboot the IP phone.
Configuring Advanced Features provides remote offices or individual users with secure access to their organization's network. There are two types of VPN access: remote-access VPN (connecting an individual device to a network) and site-to-site VPN (connecting two networks together). Remote-access VPN allows employees to access their company's intranet from home or outside the office, and site-to-site VPN allows employees in geographically separated offices to share one cohesive virtual network.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones To upload a TAR file and configure VPN via web user interface: 1. Click on Network->Advanced. 2. Click Browse to locate the TAR file from the local system. 3. Click Upload to upload the TAR file. The web user interface prompts the message “Import config…”. 4. In the VPN block, select the desired value from the pull-down list of Active. 5. Click Confirm to accept the change.
Configuring Advanced Features Quality of Service (QoS) is the ability to provide different priorities for different packets in the network, allowing the transport of traffic with special requirements. QoS guarantees are important for applications that require fixed bit rate and are delay sensitive when the network capacity is insufficient. There are four major QoS factors to be considered when configuring a modern QoS implementation: bandwidth, delay, jitter and loss.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones dropped due to interference from other lower priority traffic. VoIP can guarantee high-quality QoS only if the voice and the SIP packets are given priority over other kinds of network traffic. IP phones support the DiffServ model of QoS. Voice QoS In order to make VoIP transmissions intelligible to receivers, voice packets should not be dropped, excessively delayed, or made to suffer varying delay.
Configuring Advanced Features 3. Enter the desired value in the SIP QoS (0~63) field. 4. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. 5. Click OK to reboot the IP phone. Network Address Translation (NAT) is essentially a translation table that maps public IP address and port combinations to private ones. This reduces the need for a large number of public IP addresses.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones assistance from a third-party network server (STUN server) usually located on public Internet. The IP phone can be configured to act as a STUN client, to send exploratory STUN messages to the STUN server. The STUN server uses those messages to determine the public IP address and port used, and then informs the client. The NAT traversal and STUN server are configurable on a per-line basis.
Configuring Advanced Features IEEE 802.1X authentication is an IEEE standard for Port-based Network Access Control (PNAC), part of the IEEE 802.1 group of networking protocols. It offers an authentication mechanism for devices to connect/link to a LAN or WLAN. The 802.1X authentication involves three parties: a supplicant, an authenticator and an authentication server. The supplicant is the IP phone that wishes to attach to the LAN or WLAN. With 802.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2) Enter the password for authentication in the MD5 Password field. b) If you select EAP-TLS: 1) Enter the user name for authentication in the Identity field. 2) Leave the MD5 Password field blank. 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system. 4) In the Device Certificates field, click Browse to select the desired client (*.pem or *.
Configuring Advanced Features 5) Click Upload to upload the certificates. c) If you select PEAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 4) Click Upload to upload the certificate. d) If you select EAP-TTLS/EAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.crt, *.cer or *.der) from your local system.
Configuring Advanced Features 4) Click Upload to upload the certificate. 3. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. 4. Click OK to reboot the IP phone. To configure the 802.1X authentication via phone user interface after: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->802.1x Settings. 2. Press or , or the Switch soft key to select the desired value from the 802.1x Mode field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2) Enter the password for authentication in the MD5 Password field. 3. Click Save to accept the change. The IP phone reboots automatically to make the settings effective after a period of time. TR-069 is a technical specification defined by the Broadband Forum, which defines a mechanism that encompasses secure auto-configuration of a CPE (Customer-Premises Equipment), and incorporates other CPE management functions into a common framework.
Configuring Advanced Features RPC Method Description File types supported by IP phones are: Firmware Image Configuration File This method is used to cause the CPE to upload a specified file to the designated location. File types supported by IP phones are: Upload Configuration File Log File This method is used to request the CPE to schedule a ScheduleInform one-time Inform method call (separate from its periodic Inform method calls) sometime in the future.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 4. Enter the URL of the ACS in the ACS URL field. 5. Select the desired value from the pull-down list of Enable Periodic Inform. 6. Enter the desired time in the Periodic Inform Interval (seconds) field. 7. Enter the user name and password authenticated by the IP phone in the Connection Request Username and Connection Request Password fields. 8. Click Confirm to accept the change.
Configuring Advanced Features Procedure IPv6 can be configured using the configuration files or locally. Configure the IPv6 address Configuration File .cfg assignment method. For more information, refer to IPv6 on page 372. Configure the IPv6 address assignment method. Local Web User Interface Navigate to: http:///servl et?p=network&q=load To configure IPv6 address assignment method via web user interface: 1. Click on Network->Basic. 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones - (Optional.) If you mark the DHCP radio box, you can configure the static DNS address in the corresponding fields. 4. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after a reboot. 5. Click OK to reboot the IP phone. To configure IPv6 address assignment method via phone user interface: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN Port. 2.
Configuring Audio Features This chapter provides information for making configuration changes for the following audio features: Headset Prior Dual Headset Audio Codecs Acoustic Clarity Technology Headset prior allows users to use headset preferentially if a headset is physically connected to the IP phone. This feature is especially useful for permanent or full-time headset users. Procedure Headset prior can be configured using the configuration files or locally. Configure headset prior.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. Select the desired value from the pull-down list of Headset Prior. 3. Click Confirm to accept the change. Dual headset allows users to use two headsets on one IP phone. To use this feature, users need to physically connect two headsets to the headset and handset jacks respectively.
Configuring Audio Features 2. Select the desired value from the pull-down list of Dual-Headset. 3. Click Confirm to accept the change. CODEC is an abbreviation of COmpress-DECompress, capable of coding or decoding a digital data stream or signal by implementing an algorithm. The object of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining the quality. This can effectively reduce the frame size and the bandwidth required for audio transmission.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The corresponding attributes of the codec are listed as follows: Codec PCMU PCMA G729 G722 G723_53 G723_63 G726-16 G726-24 G726-32 G726-40 iLBC Configuration Methods Configuration Files Web User Interface Configuration Files Web User Interface Configuration Files Web User Interface Configuration Files Web User Interface Configuration Files Web User Interface Configuration Files Web User Interface Configuration Files Web User Interface Con
Configuring Audio Features Procedure Configuration changes can be performed using the configuration files or locally. Configure the codecs to use on a per-line basis. Configure the priority and rtpmap for the enabled codec. Configuration File .cfg For more information, refer to Audio Codecs on page 377. Configure the ptime. For more information, refer to Audio Codecs on page 377. Configure the codecs to use and adjust the priority of the enabled codecs on a per-line basis.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 7. To adjust the priority of codecs, select the desired codec and then click or 8. . Click Confirm to accept the change. To configure the ptime on a per-line basis via web user interface: 204 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3. Click on Advanced. 4. Select the desired value from the pull-down list of PTime (ms). 5. Click Confirm to accept the change.
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. IP phones employ advanced AEC for hands-free operation. Echo cancellation is achieved using the echo canceller. Procedure AEC can be configured using the configuration files or locally. Configure AEC.
Administrator’s Guide for SIP-T2xP and SIP-T19P 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 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 in a conversation. It is a part of the silence suppression or VAD handling for VoIP technology. CNG, in conjunction with VAD algorithms, quickly responds when periods of silence occur and inserts artificial noise until voice activity resumes.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in even intervals. Jitter is a term indicating variations in packet arrival time, which can occur because of network congestion, timing drift or route changes.
Configuring Audio Features 5. Enter the fixed delay time for fixed jitter buffer in the Normal field. Valid values range from 0 to 300. 6. Click Confirm to accept the change.
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Configuring Security Features This chapter provides information for making configuration changes for the following security-related features: Transport Layer Security Secure Real-Time Transport Protocol Encrypting Configuration Files TLS is a commonly-used protocol for providing communications privacy and managing the security of message transmission, allowing IP phones to communicate with other remote parties and connect to the HTTPS URL for provisioning in a way that is designed to prevent ea
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones AES256-SHA EDH-RSA-DES-CBC3-SHA EDH-DSS-DES-CBC3-SHA DES-CBC3-SHA DHE-RSA-AES128-SHA DHE-DSS-AES128-SHA AES128-SHA IDEA-CBC-SHA DHE-DSS-RC4-SHA RC4-SHA RC4-MD5 EXP1024-DHE-DSS-DES-CBC-SHA EXP1024-DES-CBC-SHA EDH-RSA-DES-CBC-SHA EDH-DSS-DES-CBC-SHA DES-CBC-SHA EXP1024-DHE-DSS-RC4-SHA EXP1024-RC4-SHA EXP1024-RC4-MD5 EXP-EDH-RSA-DES-CBC-SHA EXP-EDH-DSS-DES-CBC-SHA EX
Configuring Security Features negotiation with “Server Hello Done” message. Step3: IP phone sends session key information (encrypted by server’s public key) in the “Client Key Exchange” message. Step4: Server sends “Change Cipher Spec” message to activate the negotiated options for all future messages it will send. IP phones can encrypt SIP with TLS, which is called SIPS.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Common Name Validation feature enables the IP phone to mandatorily validate the common name of the certificate sent by the connecting server. Note In TLS feature, we use the terms trusted and server certificate. These are also known as CA and device certificates. Firmware upgrade from version 71 to 72 will result in update of the default server certificates. We strongly recommend that you do not downgrade the firmware.
Configuring Security Features et?p=trusted-cert&q=load Configure server certificates feature. Upload the server certificates. Navigate to: http:///servl et?p=server-cert&q=load To configure TLS on a per-line basis via web user interface: 1. Click on Account->Register. 2. Select the desired account from the pull-down list of Account. 3. Select TLS from the pull-down list of Transport. 4. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. Select the desired values from the pull-down lists of Only Accept Trusted Certificates, Common Name Validation and CA Certificates. 3. Click Confirm to accept the change. To upload a trusted certificate via web user interface: 1. Click on Security->Trusted Certificates. 2. Click Browse to select the certificate (*.pem, *.crt, *.cer or *.der) from your local system. 3. Click Upload to upload the certificate.
Configuring Security Features 2. Select the desired value from the pull-down list of Device Certificates. 3. Click Confirm to accept the change. To upload a server certificate via web user interface: 1. Click on Security->Server Certificates. 2. Click Browse to select the certificate (*.pem and *.cer) from your local system. 3. Click Upload to upload the certificate. A dialog box pops up to prompt “Success: The Server Certificate has been loaded! Rebooting, please wait…”.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:NzkyM2FjNzQ2ZDgxYjg0MzQwMGVmMGUxMzdmNWFm a=crypto:3 F8_128_HMAC_SHA1_80 inline:NDliMWIzZGE1ZTAwZjA5ZGFhNjQ5YmEANTMzYzA0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv The callee receives the INVITE message with the RTP encryption algorithm, and then answers the call by responding
Configuring Security Features Configure SRTP feature on a per-line basis. Local Web User Interface Navigate to: http:///servlet ?p=account-adv&q=load&acc= 0 To configure SRTP feature via web user interface: 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3. Click on Advanced. 4. Select the desired value from the pull-down list of RTP Encryption (SRTP). 5. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones symmetric keys (the same or different keys for configuration files) and generates encrypted configuration files with the same file name as before. This tool also encrypts the plaintext 16-character symmetric keys using a fixed key, which is the same as the one built in the IP phone, and generates new files named as .enc (xx indicates the name of the configuration file, for example, y000000000000_Security.enc for y000000000000.cfg file).
Configuring Security Features automatically in the directory where the application tool is located. 2. Click Browse to locate configuration file(s) (e.g., y000000000000.cfg) from your local system in the Select File(s) field. To select multiple configuration files, you can select the first file and then press and hold the Ctrl key and select the next files. 3. (Optional.) Click Browse to locate the target directory from your local system in the Target Directory field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The target directory will be automatically opened. You can find the encrypted CFG file(s), encrypted key file(s) and an Aeskey.txt file storing plaintext AES key(s). Procedure Decryption method can be configured using the configuration files. Configure the decryption method. Configuration File .cfg Configure AES keys. For more information, refer to Configuring Decryption Method on page 387. Configure AES keys.
Configuring Security Features 2. Enter the values in the Common AES Key and MAC-Oriented AES Key fields. AES keys must be 16 characters and the supported characters contain: 0-9, A-Z, a-z. 3. Click Confirm to accept the change.
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Upgrading Firmware This chapter provides information about upgrading the IP phone firmware. Two methods of firmware upgrade: Manually, from the local system. Automatically, from the provisioning server. The following table lists the associated firmware name for each IP phone model (X is replaced by the actual firmware version). Note IP Phone Model Associated Firmware Name SIP-T28P 2.x.x.x.rom SIP-T26P 6.x.x.x.rom SIP-T22P 7.x.x.x.rom SIP-T21P 34.x.x.x.rom SIP-T20P 9.x.x.x.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones A dialog box pops up to prompt “Firmware of the SIP Phone will be updated. It will take 5 minutes to complete. Please don't power off!”. 5. Note Click OK to confirm the upgrade. Do not unplug the network and power cables when the IP phone is upgrading firmware. Do not close and refresh the browser when the IP phone is upgrading firmware via web user interface.
Upgrading Firmware For more information, refer to Upgrading Firmware on page 389. Configure the way for the IP phone to check for Local Web User Interface configuration files. Navigate to: http:///servl et?p=settings-autop&q=load To configure the way for the IP phone to check for new configuration files via web user interface: 1. Click on Settings->Auto Provision. 2. Make the desired change. 3. Click Confirm to accept the change.
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Resource Files When configuring particular features, you may need to upload resource files (e.g., local contact directory, remote phone book) to IP phones. The resources files can be local contact directory, remote phone book and so on. Ask Yealink field application engineer for resource file templates. If the resource file is to be used for all IP phones of the same model, the resource file access URL is best specified in the .cfg file.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Procedure Use the following procedures to customize a replace rule template. To customize a replace rule template: 1. Open the template file using an ASCII editor. 2. Add the following string to the template, each starting on a separate line: Where: Prefix="" specifies the numbers to be replaced. Replace="" specifies the alternate string instead of what the user enters.
Resource Files in the section Creating Dial Plan on page 32. Procedure Use the following procedures to customize a dial-now template. To customize a dial-now template: 1. Open the template file using an ASCII editor. 2. Add the following string to the template, each starting on a separate line: Where: DialNowRule="" specifies the dial-now rule. LineID="" specifies the desired line(s) for this rule.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones indicates the start of the default soft key list and indicates the end of the default soft key list, the default soft keys are displayed on the LCD screen by default. Procedure Use the following procedures to customize a softkey layout template. To customize a softkey layout template: 1. Open the template file using an ASCII editor. 2. For each soft key that you want to enable, add the following string to the file.
Resource Files You can add contacts one by one on the IP phone directly. You can also add multiple contacts at a time and/or share contacts between IP phones using the local contact template file. After setup, place the template file to the provisioning server and specify the access URL of the template file in the configuration files. When editing a local contact template, learn the following: indicates the start of a contact list and indicates the end of a contact list.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones mobile_number="" specifies the mobile number of the contact. other_number="" specifies the other number of the contact. line="" specifies the line you want to add this contact to. ring="" specifies the ring tone for this contact. group_id_name="" specifies the existing group you want to add the contact to. 4. Specify the values within double quotes. 5. Place this file to the provisioning server.
Resource Files 2. For each contact that you want to add, add the following strings to the phone book. Each starts on a separate line: Mary 1001 Where: Specify the contact name between and . Specify the contact number between and . 3. Specify the values within double quotes. 4. Place this file to the provisioning server.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Create directory between and . When specifying the display name of the directory list, the valid values are Local Contacts, History, Remote Phone Book (not applicable to SIP-T20P IP phones) and LDAP (not applicable to SIP-T19P and SIP-T20P IP phones). When specifying the display priority of the directory list, the valid values are 1, 2, 3 and 4. 1 is the highest priority, 4 is the lowest.
Resource Files The super search template allows you to search for a contact in your desired lists when the phone is in the dialing screen. The lists may contain Local Directory, History, Remote Phone Book and LDAP. After setup, place the super search template to the provisioning server and specify the access URL in the configuration files.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The following is an example of a super search template:
Resource Files Configure the access URL of the directory template. Configuration File .cfg For more information, refer to Access URL of Directory Template on page 396. Configure the access URL of the super search template. Configuration File .cfg For more information, refer to Access URL of Super Search Template on page 397.
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Troubleshooting This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using IP phones. IP phones can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which can help an administrator more easily find the system problem and fix it. The following are helpful for better understanding and resolving the working status of the IP phone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. Select the desired level from the pull-down list of System Log Level. 3. Click Confirm to accept the change. A dialog box pops up to prompt “Do you want to restart your machine?”. The configuration will take effect after a reboot. 4. Click OK to reboot the IP phone. After a reboot, the system log level is set as 6, the administrator debug level. Note Administrator level debugging may make some sensitive information accessible (e.g.
Troubleshooting 4. Click Confirm to accept the change. A dialog box pops up to prompt “Do you want to restart your machine?”. The configuration will take effect after a reboot. 5. Click OK to reboot the IP phone. The system log will be exported successfully to the desired syslog server after a reboot. 6. Reproduce the issue. To export a log file to the local system via web user interface: 1. Click on Settings->Configuration. 2. Mark the Local radio box in the Export System Log field. 3.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The following figure shows a portion of a log file: You can capture packet in two ways: capturing the packet via web user interface or using the Ethernet software. You can analyze the packet captured for troubleshooting purpose. To capture packets via web user interface: 244 1. Click on Settings->Configuration. 2. Click Start to start capturing signal traffic. 3. Reproduce the issue to get stack traces. 4. Click Stop to stop capturing.
Troubleshooting 5. Click Export to open the file download window, and then save the file to your local system. To capture packets using the Ethernet software: Connect the Internet port of the IP phone and the PC to the same HUB, and then use Sniffer, Ethereal or Wireshark software to capture the signal traffic.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. Select the desired value from the pull-down list of Watch Dog. 3. Click Confirm to accept the change. Status indicators may consist of the power LED, MESSAGE key LED, line key indicator, headset key indicator and the on-screen icon.
Troubleshooting 2. In the Export or Import Configuration block, click Export to open the file download window, and then save the file to your local system. This section describes solutions to common issues that may occur while using the IP phone. Upon encountering a scenario not listed in this section, contact your Yealink reseller for further support. Do one of the following: Ensure that the IP phone is properly plugged into a functional AC outlet.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones “ ” The LCD screen prompts “No Service” message when there is no available SIP account on the IP phone. Do one of the following: Ensure that an account is actively registered on the IP phone at the path Menu->Status->More->Accounts. Ensure that the SIP account parameters have been configured correctly. Press the OK key when the IP phone is idle to check the basic information (e.g., IP address, MAC address and firmware version).
Troubleshooting jitter, due to message recombination of transmission or receiving equipment (e.g., timeout handling, retransmission mechanism, buffer under run). Noisy equipment, such as a computer or a fan, may cause voice interference. Turn off any noisy equipment. Line issues can also cause this problem; disconnect the old line and redial the call to ensure another line may provide better connection.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The IP phone only uses logo file in DOB format, as the DOB format file has a high compression ratio (the size of the uncompressed file compared to that of the compressed file) and can be stored in smaller space. Tools for converting BMP format to DOB format are available. For more information, refer to Yealink_SIP-T2 Series_T19P_T4_Series_IP_Phones_Auto_Provisioning_Guide.
Troubleshooting ’ Do one of the following: Ensure that the configuration is set correctly. Reboot the IP phone. Some configurations require a reboot to take effect. Ensure that the configuration is applicable to the IP phone model. The configuration may depend on support from a server. “ ” “ ” They are codes that the IP phone sends to the server when a certain action takes place.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2. Click Reset to Factory Reset in the Reset to Factory Setting field. The web user interface prompts the message “Do you want to reset to factory?”. 3. Click OK to confirm the resetting. The phone will be reset to factory sucessfully after startup. Note Reset of your phone may take a few minutes. Do not power off until the phone starts up successfully. Factory reset can restore the original password.
Troubleshooting Phone Model SIP-T21P LCD Logo Line Memory Display Key Key 2 / Support 2 / / / / Support 132*64 132*64 pixel pixel SMS XML Browser Support 3-line (2*15 SIP-T20P characte rs and Text log Support (Non UI) an icon line) SIP-T19P 132*64 132*64 pixel pixel Support 253
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Appendix 802.1x--an IEEE Standard for port-based Network Access Control (PNAC). It is a part of the IEEE 802.1 group of networking protocols. It provides an authentication mechanism to devices wishing to attach to a LAN or WLAN. ACD (Automatic Call Distribution)--used to distribute calls from large volumes of incoming calls to the registered IP phone users. ACS (Auto Configuration server)--responsible for auto-configuration of the Central Processing Element (CPE).
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones IEEE (Institute of Electrical and Electronics Engineers)--a non-profit professional association headquartered in New York City that is dedicated to advancing technological innovation and excellence. LAN (Local Area Network)--used to interconnects network devices in a limited area such as a home, school, computer laboratory, or office building.
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 Chile(Easter Islands) −06:00 Mexico(Mexico City,
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Time Zone 258 Time Zone Name 0 United Kingdom(London) 0 Morocco +01:00 Albania(Tirane) +01:00 Austria(Vienna) +01:00 Belgium(Brussels) +01:00 Caicos +01:00 Chad +01:00 Spain(Madrid) +01:00 Croatia(Zagreb) +01:00 Czech Republic(Prague) +01:00 Denmark(Kopenhagen) +01:00 France(Paris) +01:00 Germany(Berlin) +01:00 Hungary(Budapest) +01:00 Italy(Rome) +01:00 Luxembourg(Luxembourg) +01:00 Macedonia(Skopje) +01:00 Net
Appendix Time Zone Time Zone Name +04:30 Afghanistan +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
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones This appendix describes configuration parameters in the configuration files for each feature. The configuration files are .cfg and .cfg. You can set parameters in the configuration files to configure IP phones. The .cfg and .cfg files are stored on the provisioning server. The IP phone checks for configuration files and looks for resource files when restarting the IP phone. The .
Appendix parameter “network.internet_port.type” is set to 0 (DHCP). Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example network.static_dns_enable= 0 Parameter- Configuration File network.internet_port.type .cfg Configures the Internet port type. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Format Integer Default Value 0 Valid values are: Range 0-IPv4 1-IPv6 2-IPv4&IPv6 Example network.ip_address_mode = 0 Parameter- Configuration File network.internet_port.ip .cfg Configures the IP address when the Internet port type is configured as Static IP Address and the IP address mode is configured as IPv4 Description or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Appendix Parameter- Configuration File network.internet_port.gateway .cfg Configures the default gateway when the Internet port type is configured as Static IP Address and the IP address mode is Description configured as IPv4 or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IPv4 Address Default Value Blank Range Not Applicable Example network.internet_port.gateway = 192.168.1.254 Parameter- Configuration File network.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones effect. Format IPv4 Address Default Value Blank Range Not Applicable Example network.secondary_dns = 202.101.103.54 Parameter- Configuration File network.internet_port.type .cfg Configures the Internet port type. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 0 Valid values are: Range 0-DHCP 1-PPPoE 2-Static IP Address Example network.
Appendix Parameter- Configuration File network.pppoe.password .cfg Configures the PPPoE password when the Internet port type is configured as PPPoE and the IP address mode is configured as IPv4 or Description IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format String Default Value Blank Range String within 99 characters Example network.pppoe.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones PC Port Transmission Method Parameter- Configuration File network.pc_port.speed_duplex .cfg Configures the transmission method of PC port. Note: We recommend that you do not change Description this parameter. If you change this parameter, the IP phone will reboot to make the change take effect.
Appendix Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is not applicable to SIP-T19P and SIP-T21P IP phones. SIP-T19P and SIP-T21P IP phones only support bridge mode for PC connection. Format Integer Default Value 1 Valid values are: Range 0-Router 1-Bridge Example network.bridge_mode = 1 Parameter- Configuration File network.pc_port.ip .cfg Configures the IP address for the PC port when the PC port is configured as Router.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Example network.pc_port.mask = 255.255.255.0 Parameter- Configuration File network.pc_port.dhcp_server .cfg Enables or disables the DHCP service 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 effect. It is not applicable to SIP-T19P and SIP-T21P IP phones.
Appendix port when the PC port is configured as Router. Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is not applicable to SIP-T19P and SIP-T21P IP phones. Format IP Address Default Value 10.0.0.100 Range Not Applicable Example network.dhcp.end_ip = 10.0.0.100 Replace Rule Parameter- Configuration File dialplan.replace.prefix.X .cfg Description Configures the string you want to replace. X ranges from 1 to 100.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File dialplan.replace.line_id.X .cfg Configures the desired line to apply this replace rule. The digit 0 stands for all lines. Description X ranges from 1 to 100. Note: Multiple line IDs are separated by commas. It is not applicable to SIP-T19P IP phones.
Appendix commas. It is not applicable to SIP-T19P IP phones. Format Integer Default Value Blank (for all lines) Valid values are: Range 0 to 6 (for SIP-T28P) 0 to 3 (for SIP-T26P/T22P) 0 to 2 (for SIP-T21P/T20P) Example dialplan.dialnow.line_id.1 = 1,2 Parameter- Configuration File phone_setting.dialnow_delay .cfg Configures the delay time (in seconds) for the dial-now rule.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File dialplan.area_code.min_len .cfg Description Configures the minimum length of the entered numbers. Format Integer Default Value 1 Range 1 to 15 Example dialplan.area_code.min_len = 1 Parameter- Configuration File dialplan.area_code.max_len .cfg Configures the maximum length of the entered Description numbers. Note: The value must be larger than the minimum length.
Appendix Block Out Parameter- Configuration File dialplan.block_out.number.X .cfg Description Configures the block out numbers. X ranges from 1 to 10. Format String Default Value Blank Range String within 32 characters Example dialplan.block_out.number.1 = 1234 Parameter- Configuration File dialplan.block_out.line_id.X .cfg Configures the desired line to apply this block out rule. The digit 0 stands for all lines. Description X ranges from 1 to 10.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Valid values are: Range 0-Disabled (power indicator LED is off) 1-Enabled (power indicator LED is solid green) Example phone_setting.common_power_led_enable = 1 Parameter- Configuration File phone_setting.ring_power_led_ .cfg flash_enable Enables or disables the power indicator LED to flash when the phone receives an incoming call.
Appendix Default Value 0 Valid values are: 0-Disabled (power indicator LED does not Range flash) 1-Enabled (power indicator LED slow flashes (1000ms) green) Example phone_setting.mail_power_led_flash_enable =0 Parameter- Configuration File phone_setting.mute_power_led .cfg _flash_enable Enables or disables the power indicator LED to flash when a call is mute.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Format Boolean Default Value 0 Valid values are: 0-Disabled (power indicator LED does not Range flash) 1-Enabled (power indicator LED fast flashes (500ms) green) Example phone_setting.hold_and_held_power_led_flas h_enable = 0 Parameter- Configuration File phone_setting.talk_and_dial_p .cfg ower_led_enable Enables or disables the power indicator LED to be turned on when the phone is busy.
Appendix For SIP-T19P and SIP-T21P IP phones, it configures the LCD’s contrast of the IP phone only. Note: We recommend that you set the contrast of the LCD screen to 6 as a more comfortable level. It is only applicable to SIP-T19P, SIP-T21P and SIP-T28P IP phones, and EXP39 connected to SIP-T26P and SIP-T28P IP phones. Format Integer Default Value 6 Range 1 to 10 Example phone_setting.contrast = 6 Parameter- Configuration File phone_setting.active_backlight .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones and SIP-T21P IP phones. Format Integer Default Value 30 Valid values are: 0-Always off 1-Always on 15-15s Range 30-30s 60-60s 120-120s 300-300s 600-600s 1800-1800s Example phone_setting.backlight_time = 30 Parameter- Configuration File security.user_password .cfg Configures the password of the user for web server access. Description The IP phone uses “user” as the default user password.
Appendix administrator password. Note: IP phones support ASCII characters 32-126(0x20-0x7E) only in passwords. Format administrator username:new password Default Value admin Range String within 32 characters Example security.user_password = admin:password000 Parameter- Configuration File phone_setting.lock .cfg Configures the type of phone lock. Menu Key: The Menu soft key and MESSAGE key are locked (For SIP-T20P, the MENU key is locked).
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones If it is set to 0 (Disabled), IP phone lock feature is disabled. Format Integer Default Value 0 Valid values are: 0-Disabled Range 1-Menu Key 2-Function Keys 3-All Keys Example phone_setting.lock = 1 Parameter- Configuration File phone_setting.phone_lock.unlo .cfg ck_pin Configures a new unlock PIN. Once the IP Description phone is locked, you can use the default password “123” to unlock it.
Appendix Parameter- Configuration File local_time.manual_time_enabl .cfg e Description Configures the phone to obtain time from NTP server or manual settings. Format Integer Default Value 1 Valid values are: Range 0-Manual 1-NTP Example local_time.manual_time_enable = 1 NTP Server Parameter- Configuration File local_time.manual_ntp_srv_prior .cfg Description Enables or disables the phone to use manually configured NTP server preferentially.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Example local_time.ntp_server1 = cn.pool.ntp.org Parameter- Configuration File local_time.ntp_server2 .cfg Configures the IP address or the domain name of the secondary NTP server. If the primary NTP Description server is not configured or cannot be accessed, the IP phone will request the time and date from the secondary NTP server. Format IP Address or Domain Name Default Value cn.pool.ntp.
Appendix Parameter- Configuration File local_time.time_zone_name .cfg Configures the desired time zone name. Description For more available time zone names, refer to Appendix B: Time Zones on page 257. Format String Default Value China(Beijing) Range String within 32 characters Example local_time.time_zone_name = China(Beijing) DST Parameter- Configuration File local_time.summer_time .cfg Description Enables or disables Daylight Saving Time (DST) feature.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File local_time.start_time .cfg Configures the time to start DST. If “local_time.dst_time_type” is set to 0 (By Date), use the mapping: MM: 1=Jan, 2=Feb,…, 12=Dec DD:1=the first day in a month,…, 31= the last day in a month HH:0=1am, 1=2am,…, 23=12pm If “local_time.
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” is set to 1 (Enabled).
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File local_time.offset_time .cfg Description Configures the offset time (in minutes). Format Integer Default Value Blank Range -300 to +300 Example local_time.offset_time = 120 Time Format Parameter- Configuration File local_time.time_format .cfg Configures the time format. If it is set to 0 (12 Hour), the time display will Description use 12 hour format.
Appendix Valid values are: 0-WWW MMM DD 1-DD-MMM-YY 2-YYYY-MM-DD 3-DD/MM/YYYY 4-MM/DD/YY 5-DD MMM YYYY 6-WWW DD MMM For SIP-T20P IP phones: 7-MM DD YY 8-DD MM YY 9-YY MM DD Note: “WWW” represents the abbreviation of the week, “DD” represents a two-digit day, “MMM” represents the first three letters of the month, “YYYY” represents a four-digit year, and “YY” represents a two-digit year which is not displayed on the LCD screen of SIP-T20P IP phones. Example local_time.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones http://192.168.10.25/lang+English.txt Parameter- Configuration File lang.gui .cfg Description Configures the language used on the phone user interface. Format String Default Value English Valid values are: English Chinese_S (only applicable to SIP-T19P and SIP-T21P IP phones) Chinese_T (only applicable to SIP-T19P and SIP-T21P IP phones) Range German French Italian Portuguese Polish Spanish Turkish Example lang.
Appendix English Chinese_S (only applicable to SIP-T19P and SIP-T21P IP phones) German French (not applicable to SIP-T19P and SIP-T21P IP phones) Italian Portuguese (not applicable to SIP-T19P and SIP-T21P IP phones) Spanish (not applicable to SIP-T19P and SIP-T21P IP phones) Turkish Example lang.wui = English Parameter- Configuration File phone_setting.lcd_logo.mode .cfg Configures the logo mode of the LCD screen.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 1-System logo 2-Custom logo Note: For SIP-T28 IP phones, valid values are 1(System logo) and 2(Custom logo). For SIP-T20P IP phones, valid values are 0(Disabled) and 1(Enabled). Example phone_setting.lcd_logo.mode = 1 Parameter- Configuration File lcd_logo.url .cfg Description Configures the access URL of custom logo file. Note: It is not applicable to SIP-T20P IP phones.
Appendix If it is set to 0 (Disabled), neither “#” nor “*” can be used as a send key. If it is set to 1 (# key), the pound key is used as the send key. If it is set to 2 (* key), the asterisk key is used as the send key. Format Integer Default Value 1 Valid values are: Range 0-Disabled 1-# key 2-* key Example features.key_as_send = 1 Parameter- Configuration File features.key_tone .cfg Enables or disables the IP phone to play a Description tone when a user presses a key.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Valid values are: Range 0-Disabled 1-Enabled Example features.send_key_tone = 1 Parameter- Configuration File features.hotline_number .cfg Configures the hotline number. It configures a number that the IP phone Description automatically dials out when lifting the handset, pressing the speakerphone key or the line key. Leaving it blank disables hotline feature.
Appendix Example features.hotline_delay = 4 Parameter- Configuration File features.save_call_history .cfg Enables or disables the IP phone to save call log. Description If it is set to 0 (Disabled), the IP phone cannot log the placed calls, received calls, missed calls and the forwarded calls in the call log lists. Format Boolean Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example features.save_call_history = 1 Parameter- Configuration File account.X.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 0-Disabled 1-Enabled Example account.1.missed_calllog = 1 Parameter- Configuration File phone_setting.predial_autodial .cfg Enables or disables live dialpad feature. Description If it is set to 1 (Enabled), the IP phone will automatically dial out the entered phone number without having to press any key. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example phone_setting.
Appendix Parameter- Configuration File call_waiting.enable .cfg Enables or disables call waiting feature. If it is set to 0 (Disabled), a new incoming call is automatically rejected by the IP phone with Description a busy message while during a call. If it is set to 1 (Enabled), the LCD screen will present a new incoming call while during a call. Format Boolean Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example call_waiting.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File call_waiting.on_code .cfg Description Configures the call waiting on code to activate the server-side call waiting feature. Format String Default Value Blank Range String within 32 characters Example call_waiting.on_code = *72 Parameter- Configuration File call_waiting.off_code .
Appendix Parameter- Configuration File auto_redial.interval .cfg Configures the interval (in seconds) for the IP phone to wait between redials. Description The IP phone redials the dialed number at regular intervals till the callee answers the call. Format Integer Default Value 10 Range 1 to 300 Example auto_redial.interval = 10 Parameter- Configuration File auto_redial.times .cfg Configures the redial times for the IP phone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example account.1.auto_answer = 1 Parameter- Configuration File features.auto_answer_delay .cfg Configures the delay time (in seconds) Description before the phone automatically answers an incoming call. Format Integer Default Value 1 Range 1 to 4 Example features.auto_answer_delay = 1 Parameter- Configuration File features.
Appendix Example features.call_completion_enable = 1 Parameter- Configuration File account.X.anonymous_call .cfg Enables or disables anonymous call feature for account X. If it is set to 1 (Enabled), the IP phone will block Description its identity from showing up to the callee when placing a call. The callee’s phone LCD screen presents anonymous instead of the caller’s identity. X ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Valid values are: Range 0-Off Code 1-On Code Example account.1.send_anonymous_code = 0 Parameter- Configuration File account.X.anonymous_call_onc .cfg ode Configures the anonymous call on code to activate the server-side anonymous call feature for account X. Description X ranges from 1 to 6. Note: It works only if the parameter “account.X.send_anonymous_code” is set to 1 (Enabled).
Appendix Parameter- Configuration File account.X.reject_anonymous_c .cfg all Enables or disables anonymous call rejection feature for account X. If it is set to 1 (Enabled), the IP phone will Description automatically reject incoming calls from users enabled anonymous call feature. The anonymous user’s phone LCD screen presents “Anonymity Disallowed”. X ranges from 1 to 6. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example account.1.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones code to deactivate the server-side anonymous call rejection feature for account X. X ranges from 1 to 6. Format String Default Value Blank Range String within 32 characters Example account.1.anonymous_reject_offcode = *75 Return Message When DND Parameter- Configuration File features.dnd_refuse_code .cfg Configures a return code and reason of SIP response messages when rejecting an incoming call by DND.
Appendix Format Integer Default Value 0 Valid values are: Range 0-Phone 1-Custom Example features.dnd_mode = 0 DND in Phone Mode Parameter- Configuration File features.dnd.enable .cfg Enables or disables DND feature. Description If it is set to 1 (Enabled), the IP phone will reject incoming calls on all accounts. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example features.dnd.enable = 1 Parameter- Configuration File features.dnd.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Default Value Blank Range String within 32 characters Example features.dnd.off_code = *72 DND in Custom Mode Parameter- Configuration File account.X.dnd.enable .cfg Enables or disables DND feature for account X. Description If it is set to 1 (Enabled), the IP phone will reject incoming calls on account X. X ranges from 1 to 6. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example account.1.dnd.
Appendix Default Value Blank Range String within 32 characters Example account.1.dnd.off_code = *74 Parameter- Configuration File features.busy_tone_delay .cfg Configures a period of time (in seconds) for which the busy tone is audible on the IP phone. When one party releases the call, a busy tone Description is audible to the other party indicating that the call connection breaks. If it is set to 3 (3s), a busy tone is audible for 3 seconds on the IP phone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Valid values are: Range 404-No Found 480-Temporarily not available 486-Busy here Example features.normal_refuse_code = 486 Parameter- Configuration File phone_setting.is_deal180 .cfg Enables or disables the IP phone to deal with the 180 SIP message received after the 183 Description SIP message. If it is set to 1 (Enabled), the IP phone will resume and play the local ringback tone upon a subsequent 180 message received.
Appendix Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example sip.use_out_bound_in_dialog = 1 Parameter- Configuration File account.X.advanced.timer_t1 .cfg Configures the SIP session timer T1 (in seconds) for account X. Description T1 is an estimate of the Round Trip Time (RTT) of transactions between a SIP client and SIP server. X ranges from 1 to 6. Format Float Default Value 0.5 Range 0.5 to 10 Example account.1.advanced.timer_t1 = 0.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File account.X.advanced.timer_t4 .cfg Configures the session timer of T4 (in seconds) for account X. Description T4 represents the maximum duration a message will remain in the network. X ranges from 1 to 6. Format Float Default Value 5 Range 2.5 to 60 Example account.1.advanced.timer_t4 = 5 Parameter- Configuration File account.X.session_timer.enable .
Appendix refresh the session during a call before 1800 seconds. X ranges from 1 to 6. Format Integer Default Value 1800 Range 30 to 7200 Example account.1.session_timer.expires = 1800 Parameter- Configuration File account.X.session_timer.refresher .cfg Configures the session timer refresher for account X. If it is set to 0 (UAC), refreshing the session is Description performed by the IP phone. If it is set to 1 (UAS), refreshing the session is performed by a SIP server.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 1-Enabled Example features.play_hold_tone.enable = 1 Parameter- Configuration File features.play_hold_tone.delay .cfg Configures the interval (in seconds) at which the IP phone plays a hold tone. If it is set to 30 (30s), the IP phone will play a Description hold tone every 30 seconds when there is a hold call on the IP phone. Note: It works only if the parameter “features.play_hold_tone.enable” is set to 1 (Enabled).
Appendix Call Forward Mode Parameter- Configuration File features.fwd_mode .cfg Configures the call forward mode for the IP phone. Description If it is set to 0 (Phone), call forward feature is effective for the IP phone. If it is set to 1 (Custom), you can configure call forward feature for each account. Format Integer Default Value 0 Valid values are: Range 0-Phone 1-Custom Example features.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File forward.always.target < y0000000000xx >.cfg Description Configures the destination number of the always forward. Format String Default Value Blank Range String within 32 characters Example forward.always.target = 3601 Parameter- Configuration File forward.always.on_code < y0000000000xx >.cfg Configures the always forward on code to Description activate the server-side always forward feature.
Appendix forwarded to the destination number when the callee is busy. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example forward.busy.enable = 1 Parameter- Configuration File forward.busy.target < y0000000000xx >.cfg Description Configures the destination number of the busy forward. Format String Default Value Blank Range String within 32 characters Example forward.busy.target = 3602 Parameter- Configuration File forward.busy.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Default Value Blank Range String within 32 characters Example forward.busy.off_code = *75 No Answer Forward Parameter- Configuration File forward.no_answer.enable < y0000000000xx >.cfg Enables or disables no answer forward feature. Description If it is set to 1 (Enabled), incoming calls are forward to the destination number after a period of ring time.
Appendix Default Value 2 Range 0 to 20 Example forward.no_answer.timeout = 2 Parameter- Configuration File forward.no_answer.on_code < y0000000000xx >.cfg Configures the no answer forward on code to Description activate the server-side no answer forward feature. Format String Default Value Blank Range String within 32 characters Example forward.no_answer.on_code = *76 Parameter- Configuration File forward.no_answer.off_code < y0000000000xx >.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example account.1.always_fwd.enable = 1 Parameter- Configuration File account.X.always_fwd.target .cfg Configures the destination number of the Description always forward for account X. X ranges from 1 to 6. Format String Default Value Blank Range String within 32 characters Example account.1.always_fwd.
Appendix Format String Default Value Blank Range String within 32 characters Example account.1.busy_fwd.off_code = *73 Busy Forward Parameter- Configuration File account.X.busy_fwd.enable .cfg Enables or disables busy forward feature for account X. Description If it is set to 1 (Enabled), incoming calls to the account X are forwarded to the destination number when the callee is busy. X ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones activate the server-side busy forward feature for account X. X ranges from 1 to 6. Format String Default Value Blank Range String within 32 characters Example account.1.busy_fwd.on_code = *74 Parameter- Configuration File account.X.busy_fwd.off_code .cfg Configures the busy forward off code to Description deactivate the server-side busy forward feature for account X. X ranges from 1 to 6.
Appendix Parameter- Configuration File account.X.timeout_fwd.target .cfg Configures the destination number of the no Description answer forward for account X. X ranges from 1 to 6. Format String Default Value Blank Range String within 32 characters Example account.1.timeout_fwd.target = 3603 Parameter- Configuration File account.X.timeout_fwd.timeout .cfg Configures ring times (N) to wait before forwarding incoming calls for account X.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File account.X.timeout_fwd.off_code .cfg Configures the no answer forward off code Description to activate the server-side no answer forward feature for account X. X ranges from 1 to 6. Format String Default Value Blank Range String within 32 characters Example account.1.timeout_fwd.off_code = *77 Fwd International Parameter- Configuration File forward.international.enable .
Appendix 1-Enabled Example transfer.blind_tran_on_hook_enable = 1 Parameter- Configuration File transfer.on_hook_trans_enable .cfg Enables or disables the IP phone to complete Description the semi-attended transfer or the attended transfer through on-hook. Format Boolean Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example transfer.on_hook_trans_enable = 1 Parameter- Configuration File transfer.semi_attend_tran_enable .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones locally. If it is set to 2 (Network Conference), conferences are set up by the server. X ranges from 1 to 6. Format Integer Default Value 0 Valid values are: Range 0-Local Conference 2-Network Conference Example account.1.conf_type = 0 Parameter- Configuration File account.X.conf_uri .cfg Configures the conference URI for account X. X ranges from 1 to 6. Description Note: It works only if the parameter “account.X.
Appendix Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example transfer.tran_others_after_conf_enable = 1 Phone Basis Parameter- Configuration File features.pickup.direct_pickup_e .cfg nable Enables or disables the IP phone to display the DPickup soft key when the IP phone is Description off-hook. Note: It is not applicable to SIP-T20P IP phones. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example features.pickup.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Range String within 32 characters Example features.pickup.direct_pickup_code = *97 Per-line Basis Parameter- Configuration File account.X.direct_pickup_code .cfg Configures the directed call pickup code on a per-line basis. X ranges from 1 to 6. Description Note: The directed call pickup code configured on a per-line basis takes precedence over that configured on a phone basis.
Appendix Parameter- Configuration File features.pickup.group_pickup_co .cfg de Configures the group call pickup code on a phone basis. Description Note: The group call pickup code configured on a per-line basis takes precedence over that configured on a phone basis. Format String Default Value Blank Range String within 32 characters Example features.pickup.group_pickup_code = *98 Per-line Basis Parameter- Configuration File account.X.group_pickup_code .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Note: It is not applicable to SIP-T19P IP phones. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example account.1.dialoginfo_callpickup = 1 Parameter- Configuration File wui.http_enable .cfg Enables or disables the IP phone to access its web user interface using HTTP protocol. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Appendix Range 1 to 65535 Example network.port.http = 80 Parameter- Configuration File wui.https_enable .cfg Enables or disables the IP phone to access its web user interface using HTTPS protocol. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Boolean Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example wui.https_enable = 1 Parameter- Configuration File network.port.https .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 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” header, displays “anonymity” on the callee’s phone). 2-PAI-FROM (Derives the name and number of the caller from the “PAI” header preferentially. If the server does not send the “PAI” header, derives from the “From” header).
Appendix tag in the Supported header. The caller then receives an UPDATE message from the callee, and displays the identity in the From header. X ranges from 1 to 6. Format Integer Default Value 0 Range 0 to 2 Example account.1.cp_source = 0 Parameter- Configuration File account.X.dtmf.type .cfg Configures the DTMF type for account X. If it is set to 0 (INBAND), DTMF digits are transmitted in the voice band.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File account.X.dtmf.dtmf_payload .cfg Description Configures the RFC 2833 payload type. X ranges from 1 to 6. Format Integer Default Value 101 Range 96 to 127 Example account.1.dtmf.dtmf_payload = 101 Parameter- Configuration File account.X.dtmf.info_type .cfg Configures the DTMF info type when the Description DTMF type is configured as “SIP INFO”, “AUTO or SIP INFO”. X ranges from 1 to 6.
Appendix Parameter- Configuration File features.dtmf.hide .cfg Enables or disables the IP phone to suppress Description the display of DTMF digits. If it is set to 1 (Enabled), the DTMF digits are displayed as asterisks. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example features.dtmf.hide = 1 Parameter- Configuration File features.dtmf.hide_delay .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones feature. If it is set to 0 (Disabled), the IP phone will perform the transfer as normal when pressing the transfer key during a call. If it is set to 1 (Enabled), the IP phone will transmit the specified DTMF digits to the server for completing call transfer when pressing the transfer key during a call. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example features.dtmf.
Appendix busy signal to the caller. If it is set to 1 (Enabled), the IP phone will automatically answer an incoming intercom call. Format Boolean Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example features.intercom.allow = 1 Parameter- Configuration File features.intercom.mute .cfg Enables or disables the IP phone to mute the microphone when answering an intercom call. Description If it is set to 0 (Disabled), the microphone is un-muted for incoming calls.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones play a warning tone to alert you before answering the intercom call. Format Boolean Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example features.intercom.tone = 1 Parameter- Configuration File features.intercom.barge .cfg Enables or disables the IP phone to automatically answer an incoming intercom call while there is already an active call on the IP phone.
Appendix Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example features.alert_info_tone = 1 Parameter- Configuration File account.X.alert_info_url_enable .cfg Enables or disables distinctive ring tones feature for account X. If it is set to 1 (Enabled), the IP phone will try Description to download the WAV ring tone file from the URL and then play the remote ring tone when the Alert-Info header contains a remote URL. X ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File distinctive_ring_tones.alert_info.X .cfg .ringer Configures the desired ring tones for each text. Description The value ranges from 1 to 5, the digit stands for the appropriate ring tone. X ranges from 1 to 10. Format Integer Default Value 1 Valid values are: 1-Ring1.wav Range 2-Ring2.wav 3-Ring3.wav 4-Ring4.wav 5-Ring5.wav Example distinctive_ring_tones.alert_info.1.
Appendix Germany Great Britain Greece Hungary Lithuania India Italy Japan Mexico New Zealand Netherlands Norway Portugal Spain Switzerland Sweden Russia United States Chile Czech ETSI Example voice.tone.country = Custom Parameter- Configuration File voice.tone.dial .cfg voice.tone.ring voice.tone.busy voice.tone.congestion voice.tone.callwaiting voice.tone.dialrecall voice.tone.info voice.tone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones at most four different frequencies. Duration: the time duration (in milliseconds, ranges from 0 to 30000ms) of the ring tone. You can configure at most eight different tones for one condition, and separate tones by commas (e.g., 250/200, !0/1000, 200+300/500, 600+700+800+1000/2000). The exclamation point (!) can be added optionally, which means these tones are only played once. Note: It works only if the parameter “voice.tone.
Appendix Note: It is not applicable to SIP-T20P IP phones. Format String Default Value Blank Range String within 99 characters Example remote_phonebook.data.1.name = yl01 Parameter- Configuration File remote_phonebook.display_name .cfg Configures the display name of the remote phone book. If you leave it blank, Remote Phone Book is Description displayed on the LCD screen at the path Menu->Directory. Note: It is not applicable to SIP-T20P IP phones.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File features.remote_phonebook.flash_ .cfg time Configures how often to refresh the local cache of the remote phone book. If it is set to 3600 (3600s), the IP phone will refresh the local cache of the remote phone Description book every 3600 seconds. Note: It is not applicable to SIP-T20P IP phones. Format Integer Default Value 21600 Range 3600 to 2592000 features.remote_phonebook.
Appendix stands for the entering string used as the prefix of the filter condition. Note: It is not applicable to SIP-T19P and SIP-T20P IP phones. Format String Default Value Blank Range String within 99 characters ldap.name_filter = (|(cn=%)(sn=%)) When the name prefix of the cn or sn of the Example contact record matches the search criteria, the record will be displayed on the LCD screen. Parameter- Configuration File ldap.number_filter .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File ldap.host .cfg Configures the IP address or domain name Description of the LDAP server. Note: It is not applicable to SIP-T19P and SIP-T20P IP phones. Format IP Address or Domain Name Default Value Blank Range String within 99 characters Example ldap.host = 192.168.1.20 Parameter- Configuration File ldap.port .cfg Configures the LDAP server port.
Appendix Parameter- Configuration File ldap.user .cfg Configures the user name uses to login the LDAP server. This parameter can be left blank in case the Description server allows anonymous to login. Otherwise you will need to provide the user name to access the LDAP server. Note: It is not applicable to SIP-T19P and SIP-T20P IP phones. Format String Default Value Blank Range String within 99 characters Example ldap.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones LDAP server will return all searched results. Please note that a very large value of the “Max. Hits” will slow down the LDAP search speed, therefore it should be configured according to the available bandwidth. Note: It is not applicable to SIP-T19P and SIP-T20P IP phones. Format Integer Default Value 50 Range 1 to 32000 Example ldap.max_hits = 50 Parameter- Configuration File ldap.name_attr .
Appendix Range String within 99 characters Example ldap.numb_attr = telephoneNumber Parameter- Configuration File ldap.display_name .cfg Configures the display name of the contact record displayed on the LCD screen. Description Note: It is not applicable to SIP-T19P and SIP-T20P IP phones. The value must start with “%” symbol. Format String Default Value Blank Range String within 99 characters ldap.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Note: It is not applicable to SIP-T19P and SIP-T20P IP phones. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example ldap.call_in_lookup = 1 Parameter- Configuration File ldap.ldap_sort .cfg Enables or disables the IP phone to sort the search results in alphabetical order or Description numerical order. Note: It is not applicable to SIP-T19P and SIP-T20P IP phones.
Appendix Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example features.pickup.blf_visual_enable = 1 Parameter- Configuration File features.pickup.blf_audio_enable .cfg Enables or disables the IP phone to play an alert tone when the monitored user receives Description an incoming call. Note: It is not applicable to SIP-T19P IP phones. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example features.pickup.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File account.X.music_server_uri .cfg Configures the Music on Hold server address. Examples for valid values: <10.1.3.165>, 10.1.3.165, sip:moh@sip.com, Description , or yealink.com. X ranges from 1 to 6. Note: The DNS query in this parameter only supports A query. Format String Default Value Blank Range String within 256 characters Example account.1.music_server_uri =<10.1.3.
Appendix account X. X ranges from 1 to 6. Format Boolean Default Value 0 Valid values are: Value 0-Disabled 1-Enabled Example account.1.acd.available = 1 Parameter- Configuration File acd.auto_available .cfg Enables or disables ACD auto available feature. Description If it is set to 1 (Enabled), the IP phone will automatically change the phone status to available. Format Boolean Default Value 0 Valid values are: Value 0-Disabled 1-Enabled Example acd.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File account.X.subscribe_mwi .cfg Enables or disables the IP phone to subscribe the message waiting indicator to the account for account X. Description If it is set to 1 (Enabled), the IP phone will send a SUBSCRIBE message to the server for message-summary updates. X ranges from 1 to 6. Format Boolean Default Value 0 Valid values are: Value 0-Disabled 1-Enabled Example account.1.
Appendix Parameter- Configuration File voice_mail.number.X .cfg Configures the voice mail number for Description account X. X ranges from 1 to 6. Format String Default Value Blank Value String within 99 characters Example voice_mail.number.1 = 1234 Parameter- Configuration File account.X.subscribe_mwi_to_vm .cfg Enables or disables the IP phone to subscribe the message waiting indicator to the voice mail number for account X. Description X ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Default Value G722 Valid values are: Range PCMU PCMA G729 G722 G726-16 (not applicable to SIP-T21P) G726-24 (not applicable to SIP-T21P) G726-32 G726-40 (not applicable to SIP-T21P) G723_53 Example multicast.codec = G722 Parameter- Configuration File multicast.receive_priority.enable .
Appendix 1 is the highest priority, 10 is the lowest priority. If it is set to 0, all incoming multicast paging calls will be automatically ignored. Format Integer Default Value 10 Range 0 to10 Example multicast.receive_priority.priority = 10 Parameter- Configuration File multicast.listen_address.X.label < y0000000000xx >.cfg Configures the label to be displayed on the Description LCD screen when receiving the RTP multicast. X ranges from 1 to 10.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File action_url.setup_completed .cfg action_url.registered action_url.unregistered 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.
Appendix action_url.transfer_failed Configures the URL for the predefined event. The value format is: http(s)://IP address of server/help.xml? variable name=variable value. Valid variable values are: Description $mac $ip $model $firmware $active_url $active_user $active_host $local $remote $display_local $display_remote $call_id Format URL Default Value Blank Range String within 511 characters action_url.mute = Example http://192.168.0.20/help.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones For example: 10.10.*.* stands for the IP addresses that range from 10.10.0.0 to 10.10.255.255. If left blank, the IP phone cannot receive or handle any HTTP GET request. If it is set to “any”, the IP phone will accept and handle HTTP GET requests from any IP address. Format IP Address or any Default Value Blank Range String within 511 characters Example features.action_uri_limit_ip = any Parameter- Configuration File account.X.sip_server.
Appendix Range 0 to 65535 Example account.1.sip_server.1.port = 5060 Parameter- Configuration File account.X.sip_server.Y.expires .cfg Configures the registration expires (in Description seconds) of the SIP server Y for account X. X ranges from 1 to 6. Y ranges from 1 to 2. Format Integer Default Value 3600 Range 30 to 2147483647 Example account.1.sip_server.1.expires = 3600 Parameter- Configuration File account.X.sip_server.Y.retry_counts .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Valid values are: Range 0-Concurrent registration 1-Successive registration Example account.1.fallback.redundancy_type = 0 Parameter- Configuration File account.X.fallback.timeout .cfg Configures the time interval (in seconds) for the IP phone to detect whether the working server is available by sending the Description registration request after the fallback server takes over call control.
Appendix timeout equal to the DNSTTL configured for the server that the IP phone is registered to. 2-registration: the IP phone retries to send REGISTER requests to the primary server when registration renewal. 3-duration: the IP phone retries to send requests to the primary server after the timeout defined by the account.X.sip_server.Y.failback_timeout parameter. Example account.1.sip_server.1.failback_mode = 0 Parameter- Configuration File account.X.sip_server.Y.failback_tim .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones requests to the secondary server in the failover mode. X ranges from 1 to 6. Y ranges from 1 to 2. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example account.1.sip_server.1.register_on_enable =0 Parameter- Configuration File account.X.transport .cfg Configures the transport type for account X.
Appendix query. X ranges from 1 to 6. Format Integer Default Value 0 Valid values are: Range 0-UDP 1-TCP or TLS. Example account.1.naptr_build = 0 Parameter- Configuration File network.lldp.enable .cfg Enables or disables LLDP feature on the IP phone. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Boolean Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example network.lldp.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Range 1 to 3600 Example network.lldp.packet_interval = 60 Internet Port Parameter- Configuration File network.vlan.internet_port_enable .cfg Enables or disables the IP phone to insert VLAN tag on packet from the Internet port. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Appendix Parameter- Configuration File network.vlan.internet_port_priority .cfg Configures the priority value used for passing VLAN packets. 7 is the highest priority, 0 is the lowest Description priority. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 0 Range 0 to 7 Example network.vlan.internet_port_priority = 0 PC Port Parameter- Configuration File network.vlan.pc_port_enable .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Format Integer Default Value 1 Range 1 to 4094 Example network.vlan.pc_port_vid = 1 Parameter- Configuration File network.vlan.pc_port_priority .cfg Configures the priority value used for passing VLAN packets. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 0 Range 0 to 7 Example network.vlan.
Appendix request the VLAN ID. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 132 Range 128 to 254 Example network.vlan.dhcp_option = 132 Parameter- Configuration File network.vpn_enable .cfg Enables or disables VPN feature on the IP phone. Description Note: It is not applicable to SIP-T19P IP phones. If you change this parameter, the IP phone will reboot to make the change take effect.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Example openvpn.url = http://192.168.10.25/OpenVPN.tar Parameter- Configuration File network.qos.rtptos .cfg Configures the DSCP for voice packets. The default DSCP value for RTP packets is Description 46 (Expedited Forwarding). Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 46 Range 0 to 63 Example network.qos.
Appendix Parameter- Configuration File account.X.nat.nat_traversal .cfg Enables or disables the NAT traversal for Description account X. X ranges from 1 to 6. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example account.1.nat.nat_traversal = 0 Parameter- Configuration File account.X.nat.stun_server .cfg Configures the IP address or the domain Description name of the STUN server for account X. X ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File network.802_1x.mode .cfg Configures the types of the 802.1X authentication to use on the IP phone. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 0 Valid values are: 0-Disabled Range 1-EAP-MD5 2-EAP-TLS 3-PEAP-MSCHAPv2 4-EAP-TTLS/EAP-MSCHAPv2 Example network.802_1x.
Appendix Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is only applicable to EAP-MD5, PEAP-MSCHAPv2 and EAP-TTLS/EAP-MSCHAPv2 protocols. Format String Default Value Blank Range String within 32 characters Example network.802_1x.md5_password = admin123 Parameter- Configuration File network.802_1x.root_cert_url .cfg Configures the access URL of the CA certificate used for authentication.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Format URL Default Value Blank Range String within 511 characters Example network.802_1x.client_cert_url = http://192.168.1.10/ client.pem Parameter- Configuration File managementserver.enable .cfg Description Enables or disables TR-069 feature on the IP phone. Format Integer Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example managementserver.
Appendix Format String Default Value Blank Range String within 64 characters Example managementserver.password = pwd123 Parameter- Configuration File managementserver.url .cfg Description Configures the URL of the ACS. Format URL Default Value Blank Range String within 511 characters Example managementserver.url = http://192.168.1.20/acs/ Parameter- Configuration File managementserver.connection_re .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Range Example String within 64 characters managementserver.connection_request_ password = acspwd Parameter- Configuration File managementserver.periodic_infor .cfg m_enable Enables or disables the IP phone to Description periodically report its configuration information to the ACS. Format Boolean Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example managementserver.
Appendix phone will reboot to make the change take effect. Format Integer Default Value 0 Valid values are: Range 0-IPv4 1-IPv6 2-IPv4&IPv6 Example network.ip_address_mode = 1 Parameter- Configuration File network.ipv6_internet_port.type .cfg Configures the IPv6 address assignment method. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Valid values are: Range 0-Disabled 1-Enabled Example network.ipv6_static_dns_enable= 0 Parameter- Configuration File network.ipv6_internet_port.ip .cfg Configures the IPv6 address when the IPv6 address assignment method is configured as Static IP Address and the IP Description address mode is configured as IPv6 or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Appendix Parameter- Configuration File network.ipv6_internet_port.gateway .cfg Configures the gateway when the IPv6 address assignment method is configured as Static IP Address and the IP Description address mode is configured as IPv6 or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IPv6 Address Default Value Blank Range Not Applicable Example network.ipv6_internet_port.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones and the IP address mode is configured as IPv6 or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IPv6 Address Default Value Blank Range Not Applicable network.ipv6_secondary_dns = Example 2026:1234:1:1:c3c7:c11c:5447:23a6 Parameter- Configuration File features.headset_prior .cfg Enables or disables headset prior feature.
Appendix headsets on one phone. When the IP phone joins in a cal, the users with the headset connected to the headset jack have a full-duplex conversation, while the users with the headset connected to the handset jack are only allowed to listen to. Note: It is not applicable to SIP-T19P and SIP-T21P IP phones. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example features.headset_training = 1 Parameter- Configuration File account.X.codec.Y.enable .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones For SIP-T19P/T21P IP phones: When Y=1, the default value is 1; When Y=2, the default value is 1; When Y=3, the default value is 0; When Y=4, the default value is 0; When Y=5, the default value is 1; When Y=6, the default value is 1; When Y=7, the default value is 0; When Y=8, the default value is 0. Valid values are: Range 0-Disabled 1-Enabled Example account.1.codec.1.enable = 1 Parameter- Configuration File account.X.codec.Y.
Appendix When Y=4, the default value is G723_63; When Y=5, the default value is G729; When Y=6, the default value is G722; When Y=7, the default value is iLBC; When Y=8, the default value is G726-32. Valid values are: PCMU PCMA G729 G722 Range G723_53 G723_63 G726-16 G726-24 G726-32 G726-40 iLBC Example account.1.codec.1.payload_type = PCMU Parameter- Configuration File account.X.codec.Y.priority .cfg Configures the priority for the codec. Description X ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones When Y=10, the default value is 0; When Y=11, the default value is 0. For SIP-T19P/T21P IP phones: When Y=1, the default value is 1; When Y=2, the default value is 2; When Y=3, the default value is 0; When Y=4, the default value is 0; When Y=5, the default value is 3; When Y=6, the default value is 4; When Y=7, the default value is 0; When Y=8, the default value is 0. Range 0 to 10 Example account.1.codec.1.
Appendix When Y=3, the default value is 4; When Y=4, the default value is 4; When Y=5, the default value is 18; When Y=6, the default value is 9; When Y=7, the default value is 106; When Y=8, the default value is 102. Range 0 to 127 Example account.1.codec.1.rtpmap = 0 Ptime Parameter- Configuration File account.X.ptime .cfg Configures the ptime (in milliseconds) for Description the codec. X ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File voice.vad .cfg Description Enables or disables VAD feature on the IP phone. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example voice.vad = 1 Parameter- Configuration File voice.cng .cfg Description Enables or disables CNG feature on the IP phone.
Appendix 0-Fixed 1-Adaptive Example voice.jib.adaptive = 1 Parameter- Configuration File voice.jib.min .cfg Configures the minimum delay time for jitter Description buffer. Note: It works only if the parameter “voice.jib.adaptive” is set to 1 (Adaptive). Format Integer Default Value 60 Range 0 to 400 Example voice.jib.min = 60 Parameter- Configuration File voice.jib.max .cfg Configures the maximum delay time for Description jitter buffer.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Range 0 to 400 Example voice.jib.mormal = 120 Parameter- Configuration File account.X.transport .cfg Configures the transport type for account X. If it is 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 0 (UDP) Valid values are: 0-UDP Range 1-TCP 2-TLS 3-DNS-NAPTR Example account.1.
Appendix 1-Enabled Example security.trust_certificates = 1 Parameter- Configuration File security.ca_cert .cfg Configures the type of certificates the IP phone used to authenticate the connecting Description server. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 2 Valid values are: Range 0-Default certificates 1-Custom certificates 2-All certificates Example security.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File security.dev_cert .cfg Configures the type of certificates the IP phone sends for authentication. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 0 Valid values are: Range 0-Default certificates 1-Custom certificates Example security.dev_cert = 0 Parameter- Configuration File trusted_certificates.
Appendix Default Value Blank Range String within 511 characters Example server_certificates.url = http://192.168.1.20/ca.pem Parameter- Configuration File account.X.srtp_encryption .cfg Configures whether to use voice encryption service. If it is set to 1 (Optional), the IP phone will Description negotiate with the other IP phone what type of encryption to utilize for the session. If it is set to 2 (Compulsory), the IP phone is forced to use SRTP during a call. X ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones (e.g., key1). The IP phone then decrypts the encrypted configuration files using corresponding key (e.g., key2, key3). If it is set to 0 (Disabled), the IP phone will decrypt the encrypted configuration files using plaintext AES keys configured on the IP phone. Format Boolean Default Value 0 Valid values are: Value 0-Disabled 1-Enabled Example auto_provision.aes_key_in_file = 0 Parameter- Configuration File auto_provision.aes_key_16.
Appendix Default Value Range Example Blank 16 characters and the supported characters contain: 0 ~ 9, A ~ Z, a ~ z auto_provision.aes_key_16.mac = 0123456789abmins Parameter- Configuration File auto_provision.update_file_mode .cfg Enables or disables the IP phone to update Description encrypted configuration settings only during auto provisioning. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example auto_provision.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones an auto provisioning process when powered on. Format Boolean Default Value 1 Valid values are: Range 0-Disabled 1-Enabled Example auto_provision.power_on = 1 Parameter- Configuration File auto_provision.repeat.enable < y0000000000xx >.cfg Description Enables or disables the IP phone to check new configuration repeatedly. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example auto_provision.repeat.
Appendix Parameter- Configuration File auto_provision.weekly.enable < y0000000000xx >.cfg Description Enables or disables the IP phone to check new configuration weekly. Format Boolean Default Value 0 Valid values are: Range 0-Disabled 1-Enabled Example auto_provision.weekly.enable =0 Parameter- Configuration File auto_provision.weekly.begin_time < y0000000000xx >.cfg Configures the begin time of the day for the IP phone to check new configuration Description weekly.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Example auto_provision.weekly.end_time = 21:30 Parameter- Configuration File auto_provision.weekly.dayofweek < y0000000000xx >.cfg Description Configures the days of the week for the IP phone to check new configuration weekly. Format Integer Default Value 0123456 Valid values are: 0-Sunday 1-Monday Range 2-Tuesday 3-Wednesday 4-Thursday 5-Friday 6-Saturday Example 0123456 Parameter- Configuration File dialplan_replace_rule.
Appendix Parameter- Configuration File dialplan_dialnow.url .cfg Description Configures the access URL of the dial-now template. Format URL Default Value Blank Range String within 511 characters Example dialplan_dialnow.url = http://192.168.10.25/dialnow.xml Parameter- Configuration File custom_softkey_call_failed.url .cfg Configures the access URL of the custom Description file for the soft key presented on the LCD screen when in the CallFailed state.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Default Value Blank Range String within 511 characters The following example uses HTTP to download the CallIn state file from the Example “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_call_in.url = http://10.2.8.16:8080/XMLfiles/CallIn.xml Parameter- Configuration File custom_softkey_connecting.url .
Appendix custom_softkey_dialing.url = http://10.2.8.16:8080/XMLfiles/Dialing.xml Parameter- Configuration File custom_softkey_ring_back.url .cfg Configures the access URL of the custom Description file for the soft key presented on the LCD screen when in the RingBack state. Format URL Default Value Blank Range String within 511 characters The following example uses HTTP to download the RingBack state file from the “XMLfiles” directory on provisioning Example server 10.2.8.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File local_contact.data.url .cfg Description Configures the access URL of the local contact file. Format URL Default Value Blank Range String within 511 characters Example local_contact.data.url = http://192.168.10.25/contact.xml Parameter- Configuration File remote_phonebook.data.X.url .cfg Configures the access URL of the remote Description XML phone book.
Appendix Example directory_setting.url = http://192.168.1.20/favorite_setting.xml Parameter- Configuration File super_search.url .cfg Description Configures the access URL of the super search template. Format URL Default Value Blank Range String within 511 characters Example super_search.url = http://192.168.1.20/super_search.xml Parameter- Configuration File syslog.mode .cfg Configures the syslog mode.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones files. Note: It works only if the parameter “syslog.mode” is set to 1 (Server). If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address or Domain Name Default Value Blank Range String within 99 characters Example syslog.server = 192.168.1.50 Parameter- Configuration File syslog.log_level .cfg Configures the severity level of the logs to be reported to a log file.
Appendix This section provides the DSS key parameters you can configure on IP phones. DSS key consists of memory key, line key and programable key. The following table lists the number of DSS keys you can configure for each phone model: Note Phone Model Line Key Memory Key Programable Key SIP-T28P 6 10 14 SIP-T26P 3 10 14 SIP-T22P 3 / 13 SIP-T21P 2 / 11 SIP-T20P 2 / 9 SIP-T19P / / 11 The programable key takes effect only if the IP phone is idle.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Conference Forward Transfer Hold DND Call Return SMS Directed Pickup Call Park DTMF Voice Mail Speed Dial Intercom Line BLF URL Group Listening XML Group Group Pickup Multicast Paging Record XML Browser URL Record LDAP Prefix Zero Touch ACD Local Group Custom Button Keypad Lock Directory For line keys: Valid types are: 400
Appendix DND Call Return SMS (not applicable to SIP-T20P IP phones) Directed Pickup Call Park DTMF Voice Mail Speed Dial Intercom Line BLF Group Listening XML Group (not applicable to SIP-T20P IP phones) Group Pickup Multicast Paging Record XML Browser Hot Desking URL Record LDAP (not applicable to SIP-T20P IP phones) Prefix Zero Touch ACD Local Group Custom Button Keypad Lock Directory For programab
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones XML Group (not applicable to SIP-T19P IP phones) Group Pickup XML Browser History Menu Switch Account (not applicable to SIP-T19P IP phones) New SMS (not applicable to SIP-T20P IP phones) Status Hot Desking (only applicable to SIP-T19P IP phones) LDAP (not applicable to SIP-T19P and SIP-T20P IP phones) Prefix (not applicable to SIP-T20P IP phones) Zero Touch Local Directory Local Group XML D
Appendix when x=14, the default value is 2.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 45-Local Group 47-XML Directory 49-Custom Button 50-Keypad Lock 61-Directory Example memorykey.1.type = 8 Parameter- Configuration File memorykey.X.line .cfg Parameterlinekey.X.line Parameterprogramablekey.X.line Configures the desired line to apply the key feature. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Appendix Format URL Record Multicast Paging Group Listening Local Group XML Group (not applicable to SIP-T20P) ACD Hot Desking Zero Touch URL (not applicable to SIP-T20P) Keypad Lock Directory Integer For the memory key and programable key, the default value is not applicable. Default Value For the line key, when x=1, the default value is 1. When x=2, the default value is 2. … When x=6, the default value is 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. For the programable key, x ranges from 1 to 14 (For SIP-T19P IP phones, x=1-9, 13, 14; For SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14).
Appendix Configures the desired group or remote phone book when multiple groups or remote phone books are configured on the IP phone. This parameter is only applicable to Local Group/XML Group features. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Keypad Lock Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.type Parameterprogramablekey.X.type Configures a DSS key as a keypad lock key on the IP phone. The digit 50 stands for the key type Keypad Lock. For the memory key, x ranges from 1 to 10. Description For the line key, x ranges from 1 to 6.
Appendix For the programable key, x ranges from 1 to 14 (For SIP-T19P IP phones, x=1-9, 13, 14; For SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14). Format Integer Value 5 Example memorykey.1.type = 5 Directed Call Pickup Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.type Parameterprogramablekey.X.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File memorykey.X.line .cfg Parameterlinekey.X.line Parameterprogramablekey.X.line Configures the desired line to apply the directed call pickup key. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Appendix For the line key, x ranges from 1 to 6. For the programable key, x ranges from 1 to 14 (For SIP-T19P IP phones, x=1-9, 13, 14; For SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14). Format String Range String within 99 characters Example memorykey.1.value = *971001 Group Call Pickup Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File memorykey.X.line .cfg Parameterlinekey.X.line Parameterprogramablekey.X.line Configures the desired line to apply the group call pickup key. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Appendix For the programable key, x ranges from 1 to 14 (For SIP-T19P IP phones, x=1-9, 13, 14; For SIP-T20P IP phones, x=5-12, 14; For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14). Format String Range String within 99 characters Example memorykey.1.value = *98 Call Return Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.type Parameterprogramablekey.X.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Call Park Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.type Configures a DSS key as a call park key on the IP phone. Description The digit 10 stands for the key type Call Park. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Note: It is not applicable to SIP-T19P IP phones. Format Integer Value 10 Example memorykey.1.
Appendix Parameter- Configuration File memorykey.X.value .cfg Parameterlinekey.X.value Configures the call park feature code. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Note: It is not applicable to SIP-T19P IP phones. Format String Range String within 99 characters Example memorykey.1.value = *99 Intercom Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Note: It is not applicable to SIP-T19P IP phones. Format Integer Valid values are: 1 to 6 (for SIP-T28P) 1 to 3 (for SIP-T26P/T22P) Range 1 to 2 (for SIP-T21P/T20P) 1-Line 1 2-Line 2 … 6-Line 6 Example memorykey.1.line = 1 Parameter- Configuration File memorykey.X.value .cfg Parameterlinekey.X.value Configures the intercom number. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6.
Appendix For the line key, x ranges from 1 to 6. For the programable key, x ranges from 1 to 14. (For SIP-T21P IP phones, x=1-10, 14; For SIP-T22P IP phones, x=1-10, 12-14; For SIP-T26P/T28P IP phones, x ranges from 1 to 14). Note: It is not applicable to SIP-T19P and SIP-T20P IP phones. Format Integer Value 38 Example memorykey.1.type = 38 BLF Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.type Configures a DSS key as a BLF key on the IP phone.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Valid values are: 1 to 6 (for SIP-T28P) 1 to 3 (for SIP-T26P/T22P) Range 1 to 2 (for SIP-T21P/T20P) 1-Line 1 2-Line 2 … 6-Line 6 Example memorykey.1.line = 1 Parameter- Configuration File memorykey.X.value .cfg Parameterlinekey.X.value Configures the number of the monitored user. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Note: It is not applicable to SIP-T19P IP phones.
Appendix ACD Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.type Configures a DSS key as an ACD key on the IP phone. Description The digit 42 stands for the key type ACD. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Note: It is not applicable to SIP-T19P IP phones. Format Integer Value 42 Example memorykey.1.type = 42 Multicast Paging Key Parameter- Configuration File memorykey.X.type .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Parameter- Configuration File memorykey.X.value .cfg Parameterlinekey.X.value Configures the multicast IP address and port number. For the memory key, x ranges from 1 to 10. Description For the line key, x ranges from 1 to 6. Note: It is not applicable to SIP-T19P IP phones. The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255. Format IP Address Range 224.0.0.0 to 239.255.255.255 Example memorykey.1.
Appendix URL Record Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.type Configures a DSS key as a URL record key on the IP phone. Description The digit 35 stands for the key type URL Record. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Note: It is not applicable to SIP-T19P IP phones. Format Integer Value 35 Example memorykey.1.type = 35 Parameter- Configuration File memorykey.X.value .
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Hot Desking Key Parameter- Configuration File memorykey.X.type .cfg Parameterlinekey.X.type Parameterprogramablekey.X.type Configures a DSS key as a hot desking key on the IP phone. The digit 34 stands for the key type Hot Desking. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. For the programable key, x=1-9, 13, 14.
Appendix RFC 2112—Multipart MIME RFC 2246—The TLS Protocol Version 1.0 RFC 2327—SDP: Session Description Protocol RFC 2543—SIP: Session Initiation Protocol RFC 2616—Hypertext Transfer Protocol -- HTTP/1.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones RFC 3550—RTP , RTCP, IETF RFC 3550 RFC 3556—Session Description Protocol (SDP) Bandwidth Modifiers for RTCP Bandwidth RFC 3581—An Extension to the SIP for Symmetric Response Routing RFC 3608—SIP Extension Header Field for Service Route Discovery During Registration RFC 3665—Session Initiation Protocol (SIP) Basic Call Flow Examples RFC 3666—SIP Public Switched Telephone Network (PSTN) Call Flows.
Appendix RFC 4568—Session Description Protocol (SDP) Security Descriptions for Media Streams RFC 4575—A SIP Event Package for Conference State RFC 4579—SIP Call Control - Conferencing for User Agents RFC 4662—A SIP Event Notification Extension for Resource Lists RFC 5009—P-Early-Media Header RFC 5079—Rejecting Anonymous Requests in SIP RFC 5359—Session Initiation Protocol Service Examples RFC 5589—Session Initiation Protocol (SIP) Call Control - Transfer draft-levy-sip-di
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Method Supported NOTIFY Yes REFER Yes PRACK Yes INFO Yes MESSAGE Yes UPDATE Yes PUBLISH Yes Notes The following SIP request headers are supported: Method 426 Supported Accept Yes Alert-Info Yes Allow Yes Allow-Events Yes Authorization Yes Call-ID Yes Call-Info Yes Contact Yes Content-Length Yes Content-Type Yes CSeq Yes Diversion Yes Event Yes Expires Yes From Yes Max-Forwards Yes Min-SE Yes P-As
Appendix Method Supported P-Preferred-Identity Yes Proxy-Authenticate Yes Proxy-Authorization Yes RAck Yes Record-Route Yes Refer-To Yes Referred-By Yes Remote-Party-ID Yes Replaces Yes Require Yes Route Yes RSeq Yes Session-Expires Yes Subscription-State Yes Supported Yes To Yes User-Agent Yes Via Yes Notes The following SIP responses are supported: 1xx Response—Information Responses 1xx Response Supported 100 Trying Yes 180 Ringing Yes 181 Call Is Being Forwa
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 2xx Response Supported 200 OK Yes 202 Accepted Yes Notes In REFER transfer.
Appendix 4xx Response Supported 423 Interval Too Brief Yes 480 Temporarily Unavailable Yes 481 Call/Transaction Does Not Notes Yes Exist 482 Loop Detected Yes 483 Too Many Hops No 484 Address Incomplete Yes 485 Ambiguous No 486 Busy Here Yes 487 Request Terminated Yes 488 Not Acceptable Here Yes 491 Request Pending No 493 Undecipherable No 5xx Response—Server Failure Responses 5xx Response Supported 500 Internal Server Error Yes 501 Not Implemented Yes 502 Bad Gateway No
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones SDP Headers v—Protocol version o—Owner/creator and session identifier Supported Yes Yes a—Media attribute Yes c—Connection information Yes m—Media name and transport address Yes s—Session name Yes t—Active time Yes 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.
Appendix The following figure illustrates the scenario of a successful call. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones. The call flow scenario is as follows: 1. User A calls User B. 2. User B answers the call. 3. User B hangs up. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 100 Trying F4. 100 Trying F5. 180 Ringing F6. 180 Ringing F7. 200 OK F8. 200 OK F9. ACK F10. ACK 2-way RTP channel established F11. BYE F12.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Step Action Description in the Request-URI field. User A is identified as the call session initiator in the From field. A unique numeric identifier is assigned to the call and is inserted in the Call-ID field. The transaction number within a single call leg is identified in the CSeq field. The media capability User A is ready to receive is specified. The port on which User B is prepared to receive the RTP data is specified.
Appendix Step Action Description response notifies User A that the connection has been made. User A sends a SIP ACK to the proxy F9 ACK—User A to Proxy Server server. The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F10 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The call flow scenario is as follows: 1. User A calls User B. 2. User B is busy on the IP phone and unable or unwilling to take another call. The call cannot be set up successfully. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 100 Trying F4. 100 Trying F5. 486 Busy Here F6. 486 Busy Here F7. ACK F8. ACK Step Action Description User A sends the INVITE message to a proxy server.
Appendix Step Action Description specified. F2 INVITE—Proxy Server to User B The proxy server maps the SIP URI in the To field to User B. Proxy server forwards the INVITE message to User B. User B sends a SIP 100 Trying response F3 100 Trying—User B to Proxy to the proxy server. The 100 Trying Server response indicates that the INVITE request has been received by User B.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The call flow scenario is as follows: 1. User A calls User B. 2. User B does not answer the call. 3. User A hangs up. The call cannot be set up successfully. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. CANCEL F6. CANCEL F7. 200 OK F8.
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 in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Step F7 Action Description User B User A wants to disconnect the call. 200 OK—User B to Proxy User B sends a SIP 200 OK response to Server the proxy server. The SIP 200 OK response indicates that User B has received the CANCEL request. F8 200 OK—Proxy Server to User The proxy server forwards the SIP 200 A OK response to notify User A that the CANCEL request has been processed successfully.
Appendix 3. User A places User B on hold. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9. INVITE B (sendonly) F10. INVITE B (sendonly) F11. 200 OK F12. 200 OK F13. ACK F14.
Administrator’s Guide for SIP-T2xP and SIP-T19P 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 in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Appendix Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server. The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The call flow scenario is as follows: 1. User A calls User B. 2. User B answers the call. 3. User C calls User B. 4. User B accepts the call from User C. Proxy Server User A User C User B F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9. INVITE A F10. INVITE A F11. 180 Ringing F12. 180 Ringing F13. INVITE B ( sendonly ) F14. INVITE B ( sendonly ) F15.
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 in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server, The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active. User C sends a SIP INVITE message to the proxy server.
Appendix Step Action Description User A sends a mid-call INVITE request F13 INVITE—User A to Proxy to the proxy server with new SDP Server session parameters, which are used to place the call on hold. F14 INVITE—Proxy Server to User The proxy server forwards the mid-call B INVITE message to User B. User B sends a 200 OK to the proxy F15 200 OK—User B to Proxy server. The 200 OK response indicates Server that the INVITE was successfully processed.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The following figure illustrates a successful call between Yealink SIP IP phones in which two parties are in a call and then one of the parties transfers the call to a third party without consultation. This is called a blind transfer. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: 446 1.
Appendix 4. User C answers the call. Call is established between User A and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9. REFER F10. 202 Accepted F11. REFER F12. 202 Accepted F17. BYE F18. BYE F19. 200 OK F20. 200 OK F21. INVITE C F22. INVITE C F23. 180 Ringing F24. 180 Ringing F25. 200 OK F26. 200 OK F27. ACK F28.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Step Action Description User A sends an INVITE message to the proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Appendix Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server, The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active. User B sends a REFER message to the F9 REFER—User B to Proxy Server proxy server.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Step Action Description requests the call. F18 INVITE—Proxy Server to User The proxy server maps the SIP URI in the C To field to User C. User C sends a SIP 180 Ringing F19 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 F20 180 Ringing—Proxy Server to Ringing response to User A.
Appendix 5. 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. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9. INVITE B （sendonly） F10. INVITE B （sendonly） F11. 200 OK F12. 200 OK F13. ACK F14. ACK F15. INVITE C F16. INVITE C F17. 180 Ringing F18. 180 Ringing F19. 200 OK F20. 200 OK F21. ACK F22. ACK 2-way RTP channel established F23. REFER F24.
Administrator’s Guide for SIP-T2xP and SIP-T19P 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 in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Appendix Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server, The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Step Action C Description sends the INVITE request to User C. User C sends a SIP 180 Ringing F17 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 F18 180 Ringing—Proxy Server to Ringing response to User A. User A User A hears the ring-back tone indicating that User C is being alerted.
Appendix Step Action Description response indicates that User B accepts the transfer. User A terminates the call session by F27 BYE—User A to Proxy Server sending a SIP BYE request to the proxy server. The BYE request indicates that User A wants to release the call. F28 BYE—Proxy Server to User B The proxy server forwards the BYE request to User B. User B sends a SIP 200 OK response to F29 200OK—User B to Proxy the proxy server.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 4. User C answers the call. Call is established between User A and User C. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 302 Move Temporarily F4. ACK F5. 302 Move Temporarily F6. ACK F7. INVITE C F8. INVITE C F9. 180 Ringing F10. 180 Ringing F11. 200 OK F12. 200 OK F13. ACK F14.
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 the User B is inserted in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Step Action Description User A sends a SIP INVITE request to the F7 INVITE—User A to Proxy Server proxy server. In the INVITE request, a unique Call-ID is generated and the Contact-URI field indicates that User A requested the call. F8 INVITE—Proxy Server to User C The proxy server maps the SIP URI in the To field to User C. The proxy server sends the SIP INVITE request to User C.
Appendix The following figure illustrates successful call forwarding between Yealink SIP IP phones in which User B has enabled busy call forward. The incoming call is forwarded to User C when 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: 1. User B enables busy call forward, and the destination number is User C. 2. User A calls User B. 3.
Administrator’s Guide for SIP-T2xP and SIP-T19P 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 in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Appendix Step Action Description ACK message. F7 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 F8 ACK—User A to Proxy Server server. The ACK message notifies the proxy server that User A has received the ACK message. User A sends a SIP INVITE request to the F9 INVITE—User A to Proxy Server proxy server.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones The following figure illustrates successful call forwarding between Yealink SIP IP phones in which User B has enabled no answer call forward. The incoming call is forwarded to 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. The call flow scenario is as follows: 1.
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 in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Step Action Description ACK message. F7 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 F8 ACK—User A to Proxy Server server. The ACK message notifies the proxy server that User A has received the ACK message. User A sends a SIP INVITE request to the F9 INVITE—User A to Proxy Server proxy server.
Appendix The following figure illustrates successful 3-way calling between Yealink IP phones in which 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. The call flow scenario is as follows: 1. User A calls User B. 2. User B answers the call. 3. User A places User B on hold. 4. User A calls User C. 5.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones 6. User A mixes the RTP channels and establishes a conference between User B and User C. User A User B Proxy Server F1. INVITE B F4. 180 Ringing F6. 200 OK F7. ACK F2. INVITE B F3. 180 Ringing F5. 200 OK F8. ACK Session1 established between User A and User B is active F9. INVITE(sendonly) Initiate three party conference F10. INVITE (sendonly) F11. 200 OK F12. 200 OK F13. ACK F14.
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 in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server. The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
Appendix Step Action C Description sends the SIP INVITE request to User C. User C sends a SIP 180 Ringing F17 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 F18 180 Ringing—Proxy Server to Ringing response to User A. User A User A hears the ring-back tone indicating that User C is being alerted.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones This section provides the sample configuration file necessary to configure the IP phone. Any line beginning 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 .cfg or .cfg file. The parameters included here are examples only.
Appendix phone_setting.dialnow_delay = dialplan.replace.prefix.1 = dialplan.replace.replace.1 = dialplan.replace.line_id.1 = dialplan.item.1 = #Time Settings local_time.time_zone = local_time.time_zone_name = local_time.ntp_server1 = local_time.ntp_server2 = local_time.interval = local_time.dhcp_time = #Use the following parameters to set the time and date manually. local_time.manual_time_enable = local_time.date_format = local_time.time_format = #Auto DST Settings local_time.summer_time = local_time.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones #Call Hold features.play_hold_tone.enable = features.play_hold_tone.delay = sip.rfc2543_hold = #Hotline features.hotline_number = features.hotline_delay = #Web Server Type wui.http_enable = network.port.http = wui.https_enable = network.port.https = #DTMF Suppression features.dtmf.hide = features.dtmf.hide_delay = #Call Forward # In Phone Mode features.fwd_mode = 0 forward.always.enable = forward.always.target = forward.always.on_code = forward.
Appendix account.1.busy_fwd.on_code = account.1.busy_fwd.off_code = account.1.timeout_fwd.enable = account.1.timeout_fwd.target = account.1.timeout_fwd.timeout = account.1.timeout_fwd.on_code = account.1.timeout_fwd.off_code = #Call Transfer transfer.semi_attend_tran_enable = transfer.blind_tran_on_hook_enable = transfer.on_hook_trans_enable = transfer.tran_others_after_conf_enable = #Call Conference account.1.conf_type = account.1.conf_uri = #DTMF account.1.dtmf.type = account.1.dtmf.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones #LDAP ldap.enable = ldap.name_filter = ldap.number_filter = ldap.host = ldap.port = ldap.base = ldap.user = ldap.password = ldap.max_hits = ldap.name_attr = ldap.numb_attr = ldap.display_name = ldap.version = ldap.call_in_lookup = ldap.ldap_sort = #Action URL action_url.setup_completed = action_url.registered = action_url.unregistered = action_url.register_failed = action_url.off_hook = action_url.on_hook = action_url.incoming_call = action_url.
Appendix action_url.busy_to_idle = action_url.idle_to_busy = action_url.ip_change = action_url.forward_incoming_call = action_url.reject_incoming_call = action_url.answer_new_incoming_call = action_url.transfer_finished = action_url.transfer_failed = #Access URL of Resource Files dialplan_dialnow.url = dialplan_replace_rule.url = local_contact.data.url = remote_phonebook.data.1.url = directory_setting.url = super_search.
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Index Numeric C 180 Ring Workaround 90 Call Completion 802.
Administrator’s Guide for SIP-T2xP and SIP-T19P IP Phones Enabling the Watch Dog Feature 245 Music on Hold 148 G N Getting Information from Status Indicators 246 NAT Traversal Getting Started Network Address Translation (NAT) 13 Group Call Pickup 110 187 Network Conference 187 105 No Answer Forward 98 H H.
Index SRTP 217 STUN Server 187 Suppress DTMF Display 124 Summary of Changes vi T Table of Contents Time and Date xi 50 Transfer on Conference Hang Up Transfer via DTMF Transport Layer Security (TLS) Troubleshooting 106 125 211 241 Troubleshooting Methods 241 Troubleshooting Solutions 247 TR-069 Device Management 194 U Upgrading Firmware 225 Use Outbound Proxy in Dialog User Agent Client (UAC) 2 User Agent Server (UAS) 2 User Password 92 45 V Verifying Startup Viewing Log Files
