Copyright Copyright © 2013 YEALINK NETWORK TECHNOLOGY Copyright © 2013 Yealink Network Technology CO., LTD. All rights reserved. No parts of this publication may be reproduced or transmitted in any form or by any means, electronic or mechanical, photocopying, recording, or otherwise, for any purpose, without the express written permission of Yealink Network Technology CO., LTD. Under the law, reproducing includes translating into another language or format.
Class B Digital Device or Peripheral 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.
GNU GPL INFORMATION Yealink SIP-T2xP 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 About This Guide This guide is intended for administrators who need to properly configure, customize, manage, and troubleshoot the IP phone system rather than the end-users. It provides details on the functionality and configuration of the IP phones. Many of the features described in this guide involve network settings, which could affect the IP phone performance in the network. So an understanding of IP networking and prior knowledge of IP telephony concepts are necessary.
Administrator’s Guide for SIP-T2xP IP Phones features on IP phones. Chapter 4, “Configuring Advanced Features” describes how to configure the advanced features on IP phones. Chapter 5, “Configuring Audio Features” describes how to configure the audio features on IP phones. 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 Appendix B: Time Zones on page 247 Changes for Release 71, Guide Version 71.120 Major updates have occurred to the following sections: Configuring DSS Key on page 376 Changes for Release 71, Guide Version 71.
Administrator’s Guide for SIP-T2xP IP Phones Live Dialpad on page 69 Auto Answer on page 73 Call Completion on page 74 Anonymous Call on page 76 Anonymous Call Rejection on page 77 Busy Tone Delay on page 85 Return Code When Refuse on page 86 Early Media on page 87 180 Ring Workaround on page 87 Use Outbound Proxy in Dialog on page 89 SIP Session Timer on page 90 Session Timer on page 91 Call Return on page 112 Transfer via DTMF on page 122 Interco
About This Guide Changes for Release 70, Guide Version 2.
Administrator’s Guide for SIP-T2xP IP Phones x
Table of Contents Table of Contents About This Guide ......................................................................... v Documentations .................................................................................................................................... v In This Guide ............................................................................................................................................ v Summary of Changes .......................................................
Administrator’s Guide for SIP-T2xP IP Phones Creating Dial Plan ................................................................................................................................ 31 Replace Rule ................................................................................................................................. 32 Dial-now ......................................................................................................................................... 33 Area Code .........
Table of Contents Group Call Pickup..............................................................................................................................106 Dialog-Info Call Pickup .....................................................................................................................110 Call Return ...........................................................................................................................................112 Call Park.....................................
Administrator’s Guide for SIP-T2xP IP Phones Headset Prior.......................................................................................................................................195 Dual Headset ......................................................................................................................................196 Audio Codecs ....................................................................................................................................
Table of Contents Why does the IP phone use DOB format logo file instead of popular BMP, JPG and so on? .......................................................................................................................................................240 How to increase or decrease the volume? ...........................................................................240 What will happen if I connect both PoE cable and power adapter? Which has the higher priority? ......................................
Administrator’s Guide for SIP-T2xP IP Phones Appendix F: Sample Configuration File .........................................................................................441 Index ........................................................................................
Product Overview Product Overview This chapter contains the following information about SIP-T2xP IP phones: VoIP Principle SIP Components SIP IP Phone Models VoIP Principle VoIP VoIP (Voice over Internet Protocol) is a technology using the Internet Protocol instead of traditional Public Switch Telephone Network (PSTN) technology for voice communications.
Administrator’s Guide for SIP-T2xP IP Phones network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control attributes of an end-to-end call. SIP provides capabilities to: Determine the location of the target endpoint -- SIP supports address resolution, name mapping, and call redirection.
Product Overview request to its destination, as the request URI always specifies the host which is essential. The port and protocol are not always specified by the request URI. Thus if the request does not specify a port or protocol, a default port or protocol is contacted.
Administrator’s Guide for SIP-T2xP IP Phones A working IP network is established. Routers are configured for VoIP. VoIP gateways are configured for SIP. The latest (or compatible) firmware of SIP-T2xP IP phones is available. A call server is active and configured to receive and send SIP messages. Physical Features of SIP-T2xP IP Phones This section lists the available physical features of SIP-T2xP IP phones.
Product Overview SIP-T26P Physical Features: - TI TITAN chipset and TI voice engine - 132x64 graphic LCD - 3 VoIP accounts, BroadSoft/Avaya/Asterisk validated - HD Voice: HD Codec, HD Handset, HD Speaker - 45 keys including 13 DSS keys - 1xRJ9 (4P4C) handset port - 1xRJ9 (4P4C) headset port - 2xRJ45 10/100M Ethernet ports - 1XRJ12 (6P6C) expansion module port - 16 LEDs: 1xpower, 3xline, 1xmessage, 1xheadset, 10xmemory - Power adapter: AC 100~240V input and DC 5V/1.
Administrator’s Guide for SIP-T2xP 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 10/100M Ethernet ports - 5 LEDs: 1xpower, 3xline, 1xmessage - Power adapter: AC 100~240V input and DC 5V/1.2A output - Power over Ethernet (IEEE 802.
Product Overview SIP-T20P Physical Features: - TI TITAN chipset and TI voice engine - 3-line LCD consists of an icon line and two 15-character lines - 2 VoIP accounts, BroadSoft/Avaya/Asterisk validated - HD Voice: HD Codec, HD Handset, HD Speaker - 31 keys including 9 function keys - 1xRJ9 (4P4C) handset port - 1xRJ9 (4P4C) headset port - 2xRJ45 10/100M Ethernet ports - 4 LEDs: 1xpower, 2xline, 1xmessage - Power adapter: AC 100~240V input and DC 5V/1.
Administrator’s Guide for SIP-T2xP IP Phones Key Features of SIP-T2xP IP Phones In addition to physical features introduced above, SIP-T2xP IP phones also support the following key features when running the latest firmware: Phone Features - Call Options: emergency call, call waiting, call hold, call mute, call forward, call transfer, call pickup, 3-way local conference. - Basic Features: DND, phone lock, auto redial, live dialpad, dial plan, hotline, caller identity, auto answer.
Product Overview Security - HTTPS (server/client) - SRTP (RFC3711) - Transport Layer Security (TLS) - VLAN (802.
Administrator’s Guide for SIP-T2xP IP Phones 10
Getting Started Getting Started This chapter provides basic information and installation instructions of SIP-T2xP 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 Connecting the IP Phones This section introduces how to install SIP-T2xP IP phones with components in packaging contents. Note 1.
Administrator’s Guide for SIP-T2xP IP Phones 1) Attach the stand: SIP-T28P/T26P SIP-T22P/T20P 2) Connect the handset and optional headset: SIP-T28P/T26P SIP-T22P/T20P 12
Getting Started 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. Connect the included or a standard Ethernet cable between the Internet port on the IP phone and the one on the wall or switch/hub device port.
Administrator’s Guide for SIP-T2xP IP Phones 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. The IP phone can also share the network with another network device such as a PC (personal computer). It is an optional connection.
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 IP Phones 2. The message “Initializing, Please Wait” appears on the LCD screen as the IP phone starts up. 3. 4. The main LCD screen displays the following: Time and date Soft key labels (not supported by the SIP-T20P IP phone) Press the OK key to check the IP phone status, the LCD screen displays the valid IP address, MAC address, firmware version, etc. If the IP phone has successfully passed through these steps, it starts up properly and is ready for use.
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 SIP-T2xP IP phone model is: SIP-T28P: y000000000000.cfg SIP-T26P: y000000000004.cfg SIP-T22P: y000000000005.cfg SIP-T20P: y000000000007.cfg In order to deploy IP phones using the configuration files (.
Administrator’s Guide for SIP-T2xP IP Phones Reading Icons Icons associated with different features may appear on the LCD screen. The following table provides a description for each icon on SIP-T2xP IP phone models.
Getting Started T28P T26P T22P T20P Description Phone Lock Received Calls Placed Calls Missed Calls / Recording box is full / A call cannot be recorded / Recording starts successfully / / Recording cannot be started Recording cannot be stopped Configuring Basic Network Parameters This section describes how to configure basic network parameters for the IP phone. Note This section mainly introduces IPv4 network parameters. IP phones also support IPv6.
Administrator’s Guide for SIP-T2xP IP Phones and other control information are carried in tagged data items that are stored in the options field of the DHCP message. The data items themselves are also called options. DHCP can be initiated by simply connecting the IP phone with the network. IP phones broadcast DISCOVER messages to request the network information carried in DHCP options, and the DHCP server responds with specific values in corresponding options.
Getting Started Procedure DHCP can be configured using the configuration files or locally. Configure DHCP on the IP phone. Configuration File .cfg For more information, refer to DHCP on page 250. Configure DHCP on the IP phone. Web User Interface Local Navigate to: http:///servlet ?p=network&q=load Phone User Interface Configure DHCP on the IP phone. To configure DHCP via web user interface: 1. Click on Network->Basic. 2.
Administrator’s Guide for SIP-T2xP IP Phones 3. Press the Save soft key to accept the change. The IP phone reboots automatically to make settings effective after a period of time. Configuring Network Parameters Manually If DHCP is disabled or IP phones cannot obtain network parameters from the DHCP server, you need to configure them manually.
Getting Started 2. Select desired value from the pull-down list of Mode (IPv4/IPv6). 3. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after reboot. 4. Click OK to reboot the IP phone. 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.
Administrator’s Guide for SIP-T2xP IP Phones To configure the IP address mode via phone user interface: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN Port. 2. Press or to select IPv4, IPv6 or IPv4&IPv6 from the IP Mode field. 3. Press the Save soft key to accept the change. The IP phone reboots automatically to make settings effective after a period of time. To configure a static IPv4 address via phone user interface: 1.
Getting Started Phone User Interface Configure PPPoE on the IP phone.
Administrator’s Guide for SIP-T2xP IP Phones To configure PPPoE via web user interface: 1. Click on Network->Basic. 2. In the IPv4 Config block, mark the PPPoE radio box. 3. Enter the user name and password in corresponding fields. 4. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after reboot. 5. Click OK to reboot the IP phone. To configure PPPoE via phone user interface: 1.
Getting Started Auto-negotiation Auto-negotiation means that two connected devices choose common transmission parameters (e.g., speed and duplex mode) to transmit voice or data over Ethernet. This process entails devices first sharing transmission capabilities and then selecting the highest performance transmission mode supported by both. You can configure the Internet port and PC port on the IP phone to automatically negotiate during the transmission.
Administrator’s Guide for SIP-T2xP IP Phones Procedure The transmission methods of Ethernet ports can be configured using the configuration files or locally. Configure the transmission methods of Ethernet ports. Configuration File .cfg For more information, refer to Internet and PC Ports Transmission Methods on page 255. Configure the transmission methods of Ethernet ports.
Getting Started Configuring PC Port Mode The PC port on the back of the IP phone is used to connect a PC, which can be configured in one of two modes: Bridge: The IP phone functions as a bridge, and the connected PC appears on the network as a stand-alone device with its own IP address. Router: The IP phone functions as a router, and provides a DHCP service to connected PC. Procedure PC port mode can be configured using the configuration files or locally. Configure the PC port mode.
Administrator’s Guide for SIP-T2xP IP Phones 3. Mark the desired radio box. If you mark the As Router radio box, you can configure the IP address for the PC port and configure DHCP for the PC attached to the PC port. 1) Enter the IP address in the IP Address field. 2) Enter subnet mask in the Subnet Mask field. 3) Select the desired value from the pull-down list of Enable DHCP Server. 4) Enter the start IP address in the Start IP Address field. 5) Enter the end IP address in the End IP Address field. 4.
Getting Started Creating Dial Plan Regular expression, often called a pattern, is an expression that specifies a set of strings. A regular expression provides a concise and flexible means to “match” (specify and recognize) strings of text, such as particular characters, words, or patterns of characters. Regular expression is used by many text editors, utilities, and programming languages to search and manipulate text based on patterns. Regular expression can be used to define IP phone dial plan.
Administrator’s Guide for SIP-T2xP IP Phones "9001$145$2". When you dial out "0012354599" on your phone, the IP phone will replace the number with "90012354599". “$1” means 3 digits in the first parenthesis, that is, “235”. “$2” means 2 digits in the second parenthesis, that is, “99”. Replace Rule Replace rule is an alternative string that replaces the numbers entered by the user. IP phones support up to 100 replace rules, which can be created either one by one or in batch using a replace rule template.
Getting Started 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 applies to all accounts on the IP phone. 5. Click Add to add the replace rule. Dial-now Dial-now is a string used to match numbers entered by the user. When entered numbers match the predefined dial-now rule, the IP phone will automatically dial out the numbers without employing the send key.
Administrator’s Guide for SIP-T2xP IP Phones Plan on page 258. 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.
Getting Started 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 Code Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country. When entered numbers match the predefined area code rule, the IP phone will automatically add the area code before the numbers and dial out. IP phones only support one area code rule.
Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http:///servlet ?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 applies to all accounts on the IP phone. 4.
Getting Started Navigate to: http:///servlet ?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 applies to all accounts on the IP phone. 4. Click Confirm to add the block out rule.
Administrator’s Guide for SIP-T2xP IP Phones Configuring Basic Features This chapter provides information for making configuration changes for the following basic features: 38 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 An
Configuring Basic Features Session Timer Call Hold Call Forward Call Transfer Network Conference Transfer on Conference Hang Up Directed Call Pickup Group Call Pickup Dialog-Info Call Pickup Call Return Call Park Web Server Type Calling Line Identification Presentation Connected Line Identification Presentation DTMF Suppress DTMF Display Transfer via DTMF Intercom Contrast Contrast determines the readability of the texts displayed on t
Administrator’s Guide for SIP-T2xP IP Phones ?p=settings-preference&q=load Phone User Interface Configure the contrast of the LCD screen. To configure contrast via web user interface: 1. Click on Settings->Preference. 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: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Phone Settings->Contrast. 2.
Configuring Basic Features Always On: Backlight is turned on permanently. 15, 30, 60 or 120: Backlight is turned off when the IP phone is inactive after a preset period of time (in seconds), but it is automatically turned on if the status of the IP phone changes or any key is pressed. The following table lists available methods and configuration options to configure the backlight of each phone model.
Administrator’s Guide for SIP-T2xP IP Phones 3. Select the desired value from the pull-down list of Backlight Time (seconds). 4. Click Confirm to accept the change. To configure backlight via phone user interface (only applicable to the SIP-T28P IP phone): 1. Press Menu->Settings->Advanced Settings (password: admin) ->Phone Settings->Backlight. 2. Press or , or the Switch soft key to select the desired level from the Backlight Intensity field. 3.
Configuring Basic Features 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. Select user from the pull-down list of User Type. 3. Enter new password in the New Password and Confirm Password fields.
Administrator’s Guide for SIP-T2xP IP Phones 264. 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. Click on Security->Password. 2. Select admin from the pull-down list of User Type. 3. Enter the current administrator password in the Old Password field. 4.
Configuring Basic Features Phone Lock Phone lock is used to lock the IP phone to prevent it from unauthorized use. Once the IP phone is locked, a user must enter the password to unlock it. IP phones offer three types of phone lock: Menu Key, Function Keys and All Keys. 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.
Administrator’s Guide for SIP-T2xP IP Phones 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 password (numeric characters) 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.
Configuring Basic Features 2. In the desired memory key (or line 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.
Administrator’s Guide for SIP-T2xP IP Phones the time zone. Daylight Saving Time Daylight Saving Time (DST) is the practice of temporary advancing clocks during the summertime so that evenings have more daylight and mornings have less. Typically clocks are adjusted forward one hour at the start of spring and backward in autumn. Many countries have used the DST at various times, details vary by location. The DST can be adjusted automatically from the time zone configuration.
Configuring Basic Features Configure the NTP server, time zone and DST. Configure the time and date manually. Web User Interface Configure the time and date formats. Navigate to: http:///servlet Local ?p=settings-datetime&q=load Configure the NTP server and time zone. Phone User Interface Configure the time and date manually. Configure the time and date formats. To configure the NTP server, time zone and DST via web user interface: 1. Click on Settings->Time & Date. 2.
Administrator’s Guide for SIP-T2xP IP Phones - Mark the DST By Week radio box in the Fixed Type field. Select the desired values from the pull-down lists of DST Start Month, DST Start Day of Week, DST Start Day of Week Last in Month, DST Stop Month, DST Stop Day of Week and DST Stop Day of Week Last in Month. Enter the desired time in the Start Hour of Day field. Enter the desired time in the End Hour of Day field. 7. Enter the desired offset time in the Offset (minutes) field. 8.
Configuring Basic Features 4. Click Confirm to accept the change. To configure the time and data 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. Select the desired value from the pull-down list of Date Format. 4. Click Confirm to accept the change. To configure the NTP server and time zone via phone user interface: 1. Press Menu->Settings->Basic Settings->Time & Date->SNTP Settings. 2.
Administrator’s Guide for SIP-T2xP IP Phones 4. Press the Save soft key to accept the change. To configure the time and date manually via phone user interface: 1. Press Menu->Settings->Basic Settings->Time & Date->Manual Settings. 2. Enter the date in the Date field. 3. Enter the time in the Time field. 4. Press the Save soft key to accept the change. To configure the time and date formats via phone user interface: 1. Press Menu->Settings->Basic Settings->Time & Date->Time & Date Format. 2.
Configuring Basic Features The following table lists available languages and associated language packs. Available Language Associated Language Pack English lang+English.txt Deutsch lang-German.txt French lang-French.txt Italian lang-Italian.txt Portuguese lang-Portuguese.txt Polish lang-Polish.txt Spanish lang-Spanish.txt Turkish lang-Turkish.txt Procedure Loading language pack can only be performed using the configuration files.
Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http:///servlet ?p=settings-preference&q=load Phone User Interface Specify the language for the phone user interface. To specify the language for the web user interface via web user interface: 1. 54 Click on Settings->Preference.
Configuring Basic Features 2. Select the desired language from the pull-down list of Language. 3. Click Confirm to accept the change. 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.
Administrator’s Guide for SIP-T2xP IP Phones Procedure The logo shown on the idle screen can be configured using the configuration files or locally. Configure the logo shown on the idle screen. Configuration File .cfg For more information, refer to Logo Customization on page 273. Configure the logo shown on the idle screen.
Configuring Basic Features To configure a text logo via web user interface (For the SIP-T20P IP phone only): 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of User Logo. 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 display alternately.
Administrator’s Guide for SIP-T2xP IP Phones Call State CallIn Default Soft Keys Optional Soft Keys Answer Empty Forward Switch Silence Reject Connecting Empty Empty Empty Switch Empty Cancel Connecting SemiAttendTrans Transfer Empty Empty Switch Empty Cancel Dialing Send Empty IME History Delete Switch Cancel Line Directory GPickup DPickup RingBack Empty Empty Empty Switch Empty CC Cancel RingBack SemiAttendTransBack Transfer Empty Empty Switch Empty CC Cancel
Configuring Basic Features Call State Default Soft Keys Hold Held Optional Soft Keys Transfer Empty Resume Switch NewCall Answer Cancel Reject Empty Empty Empty Switch Empty Answer Cancel Reject NewCall PreTrans InConference Transfer Empty IME Directory Delete Switch Cancel Send Empty Empty Empty Switch Empty Cancel InConferenceTalk Empty Empty Empty Switch Conference Cancel Conferenced Empty Empty Hold Switch Split Answer Cancel Reject Mute Procedure Soft
Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http:///servlet ?p=settings-softkey&q=load To configure softkey layout via web user interface: 1. Click on Settings->Softkey Layout. 2. Select the desired value from the pull-down list of Custom Softkey. 3. Select the desired state from the pull-down list of Call States. 4. Select the desired soft key from the Unselected Softkeys column and then click . The selected soft key appears in the Selected Softkeys column. 5.
Configuring Basic Features Procedure Key as send can be configured using the configuration files or locally. Configure the send key. Configuration File .cfg Configure send sound. For more information, refer to Key as Send on page 275. Configure the send key. Navigate to: http:///servlet Web User Interface Local ?p=features-general&q=load Configure send sound.
Administrator’s Guide for SIP-T2xP IP Phones To configure send sound via web user interface: 1. Click on Features->Audio. 2. Select the desired value from the pull-down list of Send Sound. 3. 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 Configure the hotline number. Specify the time (in seconds) the IP phone waits before Web User Interface automatically dial out the hotline number. Navigate to: http:///servlet Local ?p=features-general&q=load Configure the hotline number. Specify the time (in seconds) the Phone User Interface IP phone waits before automatically dialing out the hotline number. To configure hotline via web user interface: 1. Click on Features->General Information. 2.
Administrator’s Guide for SIP-T2xP IP Phones 3. Enter the waiting time (in seconds) in the HotLine Delay field. 4. Press the Save soft key to accept the change. Call Log Call log contains call information such as remote party identification, time and date, and call duration. 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.
Configuring Basic Features 2. Select the desired value from the pull-down list of Save Call Log. 3. Click Confirm to accept the change. 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.
Administrator’s Guide for SIP-T2xP IP Phones Configure missed call log feature. Navigate to: Local Web User Interface http:///servlet ?p=account-basic&q=load&acc =0 To configure missed call log 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 Missed Call Log. 5. Click Confirm to accept the change.
Configuring Basic Features on page 374. Add a group and a contact to the local directory. Web User Interface Local Navigate to: http:///servlet ?p=contactsbasic&q=load&num =1&group= Phone User Interface Add a group and a contact to the local directory. 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.
Administrator’s Guide for SIP-T2xP IP Phones If Auto is selected, the IP phone will use the first available account when placing calls to the contact from the local directory. 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 AddGroup soft key. 3. Enter the desired group name in the Name field. 4.
Configuring Basic Features Live Dialpad Live dialpad allows IP phones to automatically dial out the entered phone number after a specified period of time. Procedure Live dialpad can be configured using the configuration files or locally. Configure live dialpad. Configuration File .cfg For more information, refer to Live Dialpad on page 278. Configure live dialpad.
Administrator’s Guide for SIP-T2xP IP Phones enabled. Procedure Call waiting and call waiting tone can be configured using the configuration files or locally. Configure call waiting and call Configuration File .cfg waiting tone. For more information, refer to Call Waiting on page 279. 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 IP Phones Waiting field. 3. Press or , or the Switch soft key to select the desired value from the Play 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 Auto redial allows IP phones to redial a busy number after the first attempt.
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 IP Phones Configure auto answer. Navigate to: Web User Interface Local http:///servlet ?p=account-basic&q=load&acc =0 Phone User Interface Configure auto answer. To configure auto answer 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 Auto Answer. 5. Click Confirm to accept the change.
Configuring Basic Features IP phones support call completion using the SUBSCRIBE/NOTIFY method, which is specified in draft-poetzl-sipping-call-completion-00, to subscribe to the busy party and receive notifications oftheir status changes. Procedure Call completion can be configured using the configuration files or locally. Configure call completion. Configuration File .cfg For more information, refer to Call Completion on page 281. Configure call completion.
Administrator’s Guide for SIP-T2xP IP Phones Completion field. 3. Press the Save soft key to accept the change. Anonymous Call Anonymous call allows the caller to conceal the identity from the callee. The callee’s phone LCD screen prompts an incoming call from anonymity. Anonymous call is configurable on a per-line basis. Example of anonymous SIP header: Via: SIP/2.0/UDP 10.2.8.183:5063;branch=z9hG4bK1535948896 From: "Anonymous" ;tag=128043702 To:
Configuring Basic Features Phone User Interface Configure anonymous call. 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 Send Anonymous. 5. Select the desired value from the pull-down list of Anonymous Code. 6. (Optional.) Enter the anonymous call on code in the On Code field. 7. (Optional.
Administrator’s Guide for SIP-T2xP IP Phones LCD screen presents “Anonymity Disallowed”. Anonymous call rejection is configurable on a per-line basis. The anonymous call rejection on code and anonymous call rejection off code configured on IP phones are used to activate/deactivate the server-side anonymous call rejection feature. They may vary on different servers. Procedure Anonymous call rejection can be configured using the configuration files or locally. Configure anonymous call rejection.
Configuring Basic Features 6. (Optional.) Enter the anonymous call rejection off code in the Off Code field. 7. Click Confirm to accept the change. 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 field. 3. Press Anonymous Rejection field. 4. (Optional.
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Configuring Basic Features Return Message When DND This feature defines the return code and the reason of the SIP response message for the rejected incoming call when DND is enabled on the IP phone. The caller’s phone LCD screen displays the received return code. Procedure DND can be configured using the configuration files or locally. Configure DND in the custom .cfg mode. For more information, refer to Do Not Disturb on page 285. Assign a DND key.
Administrator’s Guide for SIP-T2xP IP Phones To configure a DND key via web user interface: 1. Click on DSSKey->Memory Key (or Line Key). 2. In the desired memory key (or line 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. 82 Click on Features->Forward & DND.
Configuring Basic Features 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.) Enter the DND off code in the DND Off Code field. b) If you mark the Custom radio box: 1) Select the desired account from the pull-down list of Account. 2) Mark the desired radio box in the DND Status field. 3) (Optional.
Administrator’s Guide for SIP-T2xP IP Phones 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: 84 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.
Configuring Basic Features 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.
Administrator’s Guide for SIP-T2xP IP Phones 2． Select the desired value from the pull-down list of Busy Tone Delay (Seconds). 3． Click Confirm to accept the change. Return Code When Refuse Return code when refuse defines the return code and reason of the SIP response message for the refused call. The caller’s phone LCD screen displays the reason according to the received return code.
Configuring Basic Features message when refusing a call. Navigate to: 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 Early media refers to media (e.g., audio and video) played to the caller before a SIP call is actually established.
Administrator’s Guide for SIP-T2xP IP Phones Procedure 180 ring workaround can be configured using the configuration files or locally. Configure 180 ring workaround. Configuration File .cfg For more information, refer to 180 Ring Workaround on page 289. 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: 88 1. Click on Features->General Information. 2.
Configuring Basic Features Use Outbound Proxy in Dialog An outbound proxy server can receive all initiating request messages and route them to the designated destination. If the IP phone is configured to use an outbound proxy server within a dialog, all SIP request messages from the IP phone will be forced to send to the outbound proxy server. Note To use this feature, make sure the outbound server have been correctly configured on the IP phone.
Administrator’s Guide for SIP-T2xP IP Phones 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 Timer SIP session timers T1, T2 and T4 are SIP transaction layer timers defined in RFC 3261. Timer T1 is an estimate of the Round Trip Time (RTT) of transactions between a SIP client and SIP server. Timer T2 represents the maximum retransmitting time of any SIP request message.
Configuring Basic Features 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. The default value is 5s. 7.
Administrator’s Guide for SIP-T2xP IP Phones Procedure Session timer can be configured using the configuration files or locally. Configure session timer. Configuration File .cfg For more information, refer to Session Timer on page 291. 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: 92 1. Click on Account. 2.
Configuring Basic Features Call Hold Call hold provides a service of placing an active call on hold. When a call is placed on hold, the IP phone sends an INVITE request with a HOLD SDP to the server. IP phones support two call hold methods, one is RFC 3264, which sets the “a” (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).
Administrator’s Guide for SIP-T2xP IP Phones 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: 94 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Play Hold Tone. 3. Enter the desired time in the Play Hold Tone Delay field.
Configuring Basic Features 4. Click Confirm to accept the change. Call Forward Call forward allows users to redirect an incoming call to a third party. IP phones redirect an incoming INVITE message by responding with a 302 Moved Temporarily message, which contains a Contact header with a new URI that should be tried. Three types of call forward: Always Forward -- Forward the incoming call immediately. Busy Forward -- Forward the incoming call when the callee is busy.
Administrator’s Guide for SIP-T2xP IP Phones Configure call forward. Navigate to: http:///serv let?p=features-forward&q=lo ad Web User Interface Local Configure forward international. Navigate to: http:/// servlet?p=features-general&q =load Phone User Interface Configure call forward. To configure call forward via web user interface: 1. Click on Features->Forward & DND. 2. In the Forward block, mark the desired radio box in the Mode field.
Configuring Basic Features 2) Enter the destination number you want to forward in the Target field. 3) Enter the on code and off code in the On Code and Off Code fields. 4) Select the ring time to wait before forwarding from the pull-down list of After Ring Time (only for the no answer forward). 3. Click Confirm to accept the change. To configure forward international via web user interface: 1. Click on Features->General Information. 2.
Administrator’s Guide for SIP-T2xP IP Phones 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. 2) Enter the destination number you want to forward all incoming calls to in the Forward To field. 3) (Optional.
Configuring Basic Features 2) Enter the destination number you want to forward all incoming calls to in the Forward To field. 3) (Optional.) Enter the always forward on code and off code respectively in the On Code and Off Code fields. You can also configure the always forward for all accounts. After the always forward was configured for a specific account, do the following: 1) Press or to highlight the Always field. 2) Press the All Lines soft key. The LCD screen prompts “Copy to All Lines?”.
Administrator’s Guide for SIP-T2xP IP Phones 5. Press the Save soft key to accept the change. Call Transfer Call transfer enables IP phones to transfer an existing call to another party. IP phones support call transfer using the REFER method specified in RFC 3515 and offer three types of transfer: Blind Transfer -- Transfer a call directly to another party without consulting. Blind transfer is implemented by a simple REFER method without Replaces in the Refer-To header.
Configuring Basic Features 1. Click on Features->Transfer. 2. Select the desired values from the pull-down lists of Semi-Attend Transfer, Blind Transfer On Hook and Semi Attend Transfer On Hook. 3. Click Confirm to accept the change. Network Conference Network conference, also known as centralized conference, provides users with flexibility of call with multiple participants (more than three). IP phones implement network conference using the REFER method specified in RFC 4579.
Administrator’s Guide for SIP-T2xP IP Phones 3. Click on Advanced. 4. Select Network Conference from the pull-down list of Conference Type. 5. Enter the conference URI in the Conference URI field. 6. Click Confirm to accept the change. Transfer on Conference Hang Up For local conference, all parties drop the call when the conference initiator drops the conference call.
Configuring Basic Features ?p=features-transfer&q=load 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 Directed call pickup is used for picking up an incoming call on a specific extension.
Administrator’s Guide for SIP-T2xP IP Phones Assign a directed call pickup key. For more information, refer to Directed Call Pickup Key on .cfg page 382. Configure directed call pickup feature on a phone basis. For more information, refer to Directed Call Pickup on page 305. 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.
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. 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 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 .cfg basis. For more information, refer to Group Call Pickup on page 307. Assign a group call pickup key. Configuration File For more information, refer to Group Call Pickup Key on page 383. .cfg Configure group call pickup feature on a phone basis. For more information, refer to Group Call Pickup on page 306.
Administrator’s Guide for SIP-T2xP IP Phones pull-down list of Type. 3. 108 Enter the group call pickup code in the Value field.
Configuring Basic Features 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. 3. Enter the group call pickup code in the Group Call Pickup Code field. 4. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP IP Phones 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.
Configuring Basic Features Example of the dialog-info message carried in NOTIFY message:
Administrator’s Guide for SIP-T2xP IP Phones 4. Select the desired value from the pull-down list of Dialog Info Call Pickup. 5. Click Confirm to accept the change. Call Return Call return, also known as last call return, allows users to place a call back to the last caller. Call return is implemented on IP phones using a call return key. Procedure Call return key can be configured using the configuration files or locally. Assign a call return key. Configuration File .
Configuring Basic Features 2. In the desired memory key (or line 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. 5.
Administrator’s Guide for SIP-T2xP IP Phones Phone User Interface Assign a call park key. To configure a call park key via web user interface: 1. Click on DSSKey->Memory Key (or Line Key). 2. In the desired memory key (or line 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. Select the desired line from the pull-down list of Line. 5. Click Confirm to accept the change.
Configuring Basic Features Web server type can be configured using the configuration files or locally. Configure the web access Configuration File .cfg type, HTTP port and HTTPS port. For more information, refer to Web Server Type on page 308. Configure the web access type, HTTP port and HTTPS port. Web User Interface Navigate to: http:///servl Local et?p=network-adv&q=load Phone User Interface Configure the web access type, HTTP port and HTTPS port.
Administrator’s Guide for SIP-T2xP IP Phones 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. Press or , or the Switch soft key to select the desired value from the HTTP Status field. 3. Enter the HTTP port number in the HTTP Port field. 4. Press or , or the Switch soft key to select the desired value from the HTTPS Status field. 5.
Configuring Basic Features 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 the Caller ID Source. 5. Click Confirm to accept the change. Connected Line Identification Presentation Connected line identification presentation (COLP) allows IP phones to display the identity of the callee specified for outgoing calls.
Administrator’s Guide for SIP-T2xP IP Phones DTMF DTMF (Dual Tone Multi-frequency), better known as touch-tone, is used for telecommunication signaling over analog telephone lines in the voice-frequency band. DTMF is the signal sent from the IP phone to the network, which is generated when pressing the IP phone’s keypad during a call. Each key press on the IP phone generates one sinusoidal tone of two frequencies. One is generated from a high frequency group and the other from a low frequency group.
Configuring Basic Features same codec as your voice and is audible to conversation partners. SIP INFO DTMF digits are transmitted by the SIP INFO messages when the voice stream is established after a successful SIP 200 OK-ACK message sequence. The SIP INFO message is sent along the signaling path of the call. The SIP INFO message can support transmitting DTMF digits in three ways: DTMF, DTMF-Relay and Telephone-Event.
Administrator’s Guide for SIP-T2xP IP Phones pull-down list of DTMF Info Type. 6. Enter the desired value in the DTMF Payload Type (96~127) field. 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. 120 Click on Features->General Information.
Configuring Basic Features 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 Suppress DTMF display allows IP phones to suppress the display of DTMF digits. DTMF digits are displayed as “*” on the LCD screen. Suppress DTMF display delay defines whether to display the DTMF digits for a short period of time before displaying as “*”.
Administrator’s Guide for SIP-T2xP IP Phones 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. 4. Click Confirm to accept the change. Transfer via DTMF Call transfer is implemented via DTMF on some traditional servers.
Configuring Basic Features 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. Intercom Intercom allows establishing an audio conversation directly. The IP phone can answer intercom calls automatically. This feature depends on support from a SIP server.
Administrator’s Guide for SIP-T2xP IP Phones 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. In the desired memory key (or line key) field, select Intercom from the pull-down list of Type. 3. Enter the remote extension number in the Value field. 4. Select the desired line from the pull-down list of Line. 5. Click Confirm to accept the change.
Configuring Basic Features Intercom Mute Intercom Mute allows the IP phone to mute the microphone for incoming intercom calls. Intercom Tone Intercom Tone allows the IP phone to play a warning tone before answering an intercom call. Intercom Barge Intercom Barge allows the IP phone to automatically answer an incoming intercom call while an active call is in progress. The active call will be placed on hold. Procedure Incoming intercom calls can be configured using the configuration files or locally.
Administrator’s Guide for SIP-T2xP IP Phones 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. 126 Press the Save soft key to accept the change.
Configuring Advanced Features Configuring Advanced Features This chapter provides information for making configuration changes for the following advanced features: Distinctive Ring Tones Tones Remote Phone Book LDAP Busy Lamp Field Music on Hold Automatic Call Distribution Message Waiting Indicator Multicast Paging Call Recording Hot Desking Action URL Action URI Server Redundancy LLDP VLAN VPN Quality of Service Network Address Tr
Administrator’s Guide for SIP-T2xP IP Phones phone strips out the URL and keyword parameter and maps them to the appropriate ring tone. Alert-Info headers in the following two formats: Alert-Info: http://localIP/Bellcore-drN Alert-Info: ;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). 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 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 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. Tones When receiving a message, the IP phone will play a warning tone. You can customize tones or select specialized tone sets (vary from country to country) to indicate different conditions of the IP phone. The default tones used on IP phones are the US tone sets.
Administrator’s Guide for SIP-T2xP IP Phones Chile Czech ETSI Configured tones can be heard on IP phones for the following conditions.
Configuring Advanced Features 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 Remote phone book is centrally maintained phone book, stored on the remote server. Users only need the access URL of the remote phone book. The IP phone can establish a connection with the remote server and download the phone book, and then display the phone book entries on the phone user interface.
Administrator’s Guide for SIP-T2xP IP Phones refreshes the local cache of the remote phone book. For more information, refer to Remote Phone Book on page 319. Specify the access URL of the remote phone book. Navigate to: http:///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. Specify how often the IP phone refreshes the local cache of the remote phone book.
Configuring Advanced Features 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. LDAP LDAP (Lightweight Directory Access Protocol) is an application protocol for accessing and maintaining information services for the distributed directory over an IP network.
Administrator’s Guide for SIP-T2xP IP Phones LDAP Attributes The following table lists the most common attributes used to configure the LDAP lookup on IP phones. Abbreviation Name Description gn givenName First name cn commonName sn surname dn distinguishedName dc dc - company - telephoneNumber mobile mobilephoneNumber ipPhone IPphoneNumber LDAP attribute being made up from given name joined to surname.
Configuring Advanced Features 3. Select the desired values from the corresponding pull-down list. 4. Click Confirm to accept the change. To configure an LDAP key via web user interface: 1. Click on DSSKey->Memory Key (or Line Key). 2. In the desired memory key (or line key) field, select LDAP from the pull-down list of Type. 3. Click Confirm to accept the change. To configure an LDAP key via phone user interface: 1. Press Menu->Features->DSS Keys->Memory Keys (or Line Keys). 2.
Administrator’s Guide for SIP-T2xP IP Phones Busy Lamp Field Busy Lamp Field (BLF) is used to monitor a specific user for status changes on IP phones. For example, you can configure a BLF key on a supervisor’s phone to monitor the phone user status (busy or idle). When the monitored user makes a call, a busy indicator on the supervisor’s phone shows that the user’s phone is in use. When the monitored user is idle, the supervisor presses the BLF key to dial out the phone number.
Configuring Advanced Features LED Status Description Solid green The monitored user is idle. Fast flashing red The monitored user receives an incoming call. Solid red The monitored user is busy. Slow flashing red (1s) Off The call is parked against the monitored user’s phone number. The monitored user does not exist.
Administrator’s Guide for SIP-T2xP IP Phones Assign a BLF key. Navigate to: http:///servl et?p=dsskey&q=load&model=0 Specify whether to use visual alert and audio alert for BLF pickup. Local Web User Interface Navigate to: http:///servl et?p=features-callpickup&q=lo ad Configure LED off in idle. Navigate to: http:///servl et?p=features-general&q=load Phone User Interface Assign a BLF key. To configure a BLF key via web user interface: 1.
Configuring Advanced Features 2. Select the desired value from the pull-down list of Visual Alert for BLF Pickup. 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 LED off in idle via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of LED Off in Idle. 3. Click Confirm to accept the change.
Administrator’s Guide for SIP-T2xP IP Phones 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 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.
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 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 IP Phones For more information, refer to ACD Key on page 390. Configure ACD auto available. For more information, refer to ACD on page 328. Assign an ACD key. Navigate to: http:///servlet Web User Interface Local ?p=dsskey&q=load&model=0 Configure ACD auto available. Navigate to: http:///servlet ?p=features-acd&q=load Phone User Interface Assign an ACD key. To configure an ACD key via web user interface: 1.
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 5. 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 IP Phones Procedure Configuration changes can be performed using the configuration files or locally. Configure subscribe for MWI. Configure subscribe MWI to Configuration File .cfg voice mail. For more information, refer to Message Waiting Indicator on page 328. 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 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 391. Specify a multicast codec for the IP phone to use for multicast RTP. For more information, refer to Sending RTP Stream on page 331. Assign a multicast paging key.
Configuring Advanced Features 4. Click Confirm to accept the change. 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.
Administrator’s Guide for SIP-T2xP IP Phones in progress. If the parameter is configured as disabled, all incoming multicast paging calls will be automatically ignored. If the parameter is the priority value, the incoming multicast paging calls with higher priority are automatically answered and the ones with lower priority are ignored. 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 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 IP Phones server. IP phones themselves do not have memory to store the recording, what they can do is to trigger the recording and indicate the recording status. Normally, there are 2 main methods to trigger a recording on a certain server. We call them record and URL record. Record is for the IP phone to send the server a SIP INFO message containing a specific header. URL record is for the IP phone to send an HTTP GET message containing a specific URL to the server.
Configuring Advanced Features 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 2.71.0.140 00:16:65:11:30:68\r\n If the recording is successfully started, the server will respond with a 200 OK message. Example of a 200 OK message: The recording session is successfully started.
Administrator’s Guide for SIP-T2xP IP Phones Procedure Call recording key can be configured using the configuration files or locally. Assign a record key. For more information, refer to Configuration File .cfg Record Key on page 392. Assign a URL record key. For more information, refer to URL Record Key on page 392. Assign a record key and URL record key.
Configuring Advanced Features 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. 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 Record from the Key Type field. 5. Press the Save soft key to accept the change.
Administrator’s Guide for SIP-T2xP IP Phones Hot desking key can be configured using the configuration files or locally. Assign a hot desking key. Configuration File .cfg For more information, refer to Hot Desking Key on page 393. Assign a hot desking key. Web User Interface Local Navigate to: http:///servlet ?p=dsskey&q=load&model=0 Phone User Interface Assign a hot desking key. To configure a hot desking key via web user interface: 1.
Configuring Advanced Features or HTTPS GET request. You can specify a URL that triggers a GET request when a specified event occurs. Action URL can only be triggered by the pre-defined events (e.g., log on). The valid URL format is: http(s)://IP address of the server/help.xml?. The following table lists the pre-defined events for action URL. Event Description Setup Completed When the IP phone completes startup. Registered When the IP phone successfully registers an account.
Administrator’s Guide for SIP-T2xP IP Phones Event Description Forward Incoming Call When the IP phone forwards an incoming call. Reject Incoming Call When the IP phone rejects an incoming call. Answer New-In Call When the IP phone answers a new call. Transfer Finished When the IP phone completes to transfer a call. Transfer Failed When the IP phone fails to transfer a call. Idle To Busy Busy To Idle Call Interrupt Autop Finish When the state of the IP phone changes from idle to busy.
Configuring Advanced Features Variable Value Description call. The SIP URI of the callee when the IP phone receives an incoming call. The SIP URI of the callee when the IP phone places a $remote call. The SIP URI of the caller when the IP phone receives an incoming call. The display name of the caller when the IP phone $display_local places a call. The display name of the callee when the IP phone receives an incoming call.
Administrator’s Guide for SIP-T2xP IP Phones 2. Enter the action URLs in the corresponding fields. 3. Click Confirm to accept the change. Action URI Opposite to action URL, action URI allows IP phones to interact with web server application by receiving and handling an HTTP or HTTPS GET request. When receiving a GET request, the IP phone will perform the specified action and respond with a 200 OK message. A GET request may contain variable named as “key” and variable value, separated by “=”.
Configuring Advanced Features Variable Value L1-LX Phone Action Press the line keys (For SIP-T28P, X=6, for SIP-T226/22P, X=3, for SIP-T20P, X=2). D1-D10 Press the memory keys (Only for SIP-T28/T26P). F_CONFERENCE Press the CONF key (Except for SIP-T22P). F1-F4 Press the soft keys (Except for SIP-T20P). MSG Press the MESSAGE key. HEADSET Press the HEADSET key. RD Press the RD key. UP/DOWN/LEFT/RIGHT Press the navigation keys. Reboot the IP phone.
Administrator’s Guide for SIP-T2xP IP Phones Navigate to: http:///servl et?p=features-remotecontrl&q =load Configure reboot in talking feature. Navigate to: http:///servl et?p=features-general&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 comma.
Configuring Advanced Features 2. Select the desired value from the pull-down list of Reboot In Talking. 3. Click Confirm to accept the change. Server Redundancy Server redundancy is often required in VoIP deployments to ensure continuity of phone service, for events where the server needs to be taken offline for maintenance, the server fails, or the connection between the IP phone and the server fails. Two types of redundancy are possible.
Administrator’s Guide for SIP-T2xP IP Phones and a fallback server) are configured for per line registration. Working Server: Server 1 is configured with the domain name of the working server. For example, yealink.pbx.com. DNS mechanism is used such that the working server is resolved to multiple SIP servers for failover purpose. The working server is deployed in redundant pairs, designated as primary and secondary servers.
Configuring Advanced Features unavailable, the secondary server will serve as the working server. Procedure Server redundancy can be configured using the configuration files or locally. Configure the server redundancy on the IP phone. Configuration File .cfg For more information, refer to Server Redundancy on page 335. Configure the server redundancy on the IP phone.
Administrator’s Guide for SIP-T2xP IP Phones SIP Server Domain Name Resolution If a domain name is configured for a SIP server, the IP address(es) associated with that domain name will be resolved through DNS as specified by RFC 3263. The DNS query involves NAPTR, SRV and A queries, which allows the IP phone to adapt to various deployment environments.
Configuring Advanced Features Parameters are explained in the following table: Parameter order pref flags Description Specify preferential treatment for the specific record. The order is from lowest to highest, lower order is more preferred. Specify the preference for processing multiple NAPTR records with the same order value. Lower value is more preferred. The flag “s” means to perform an SRV lookup.
Administrator’s Guide for SIP-T2xP IP Phones SRV query returns two records. The two SRV records point to different hosts and have the same priority 0. The weight of the second record is higher than the first one, so the second record will be picked first. The two records also contain a port “5060”, the IP phone uses this port. If the Target is not a numeric IP address, the IP phone performs an A query. So in this case, the IP phone uses “server1.yealink.pbx.com" and “server2.yealink.pbx.
Configuring Advanced Features Configure the transport type on the IP phone. Local Web User Interface Navigate to: http:///servl et?p=account-register&q=load &acc=0 LLDP LLDP (Linker Layer Discovery Protocol) is a vendor-neutral Link Layer protocol, which allows IP phones to receive and/or transmit device-related information to directly connected devices on the network that are also using the protocol, and store the information that is learned about other devices.
Administrator’s Guide for SIP-T2xP IP Phones TLV Type TLV Name System Name System Description Description Name assigned to the IP phone. The default value is “yealink”. Description of the IP phone. The default value is “yealink”. The supported and enabled capabilities of the IP phone. Optional TLVs System Capabilities The supported capabilities are Bridge, Telephone and Router. The enabled capabilities are Bridge and Telephone by default. Port Description Description of port that sends data unit.
Configuring Advanced Features TLV Type TLV Name Inventory – Software Revision Inventory – Serial Number Description Software revision of the IP phone. Serial number of the IP phone. Inventory – Manufacturer name of the IP phone. Manufacturer Name The default value is “yealink”. Inventory – Model Name Asset ID Model name of the IP phone. Assertion identifier of the IP phone. The default value is “asset”. Procedure LLDP can be configured using the configuration files or locally. Configure LLDP.
Administrator’s Guide for SIP-T2xP IP Phones 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 reboot. 5. Click OK to reboot the IP phone. VLAN VLAN (Virtual Local Area Network) is used to logically divide a physical network into several broadcast domains. VLAN membership can be configured through software instead of physically relocating devices or connections.
Configuring Advanced Features VLAN Discovery via DHCP IP phones support VLAN discovery via DHCP. When the VLAN Discovery method is set to DHCP, the IP phone will examine DHCP option for a valid VLAN ID. The predefined option 132 is used to supply the VLAN ID by default. You can customize the DHCP option used to request the VLAN ID. Procedure VLAN can be configured using the configuration files or locally. Configure VLAN for the Internet port and PC port manually.
Administrator’s Guide for SIP-T2xP IP Phones 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 reboot to make the settings effective. 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.
Configuring Advanced Features 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 reboot. 5. Click OK to reboot the IP phone.
Administrator’s Guide for SIP-T2xP IP Phones provides remote offices or individual users with secure access to their organization's network. 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.
Configuring Advanced Features 2. Click Browse to locate the tar file from the local system. 3. Click Import to import 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. A dialog box pops up to prompt that settings will take effect after reboot. 6. Click OK to reboot the IP phone. To configure VPN via phone user interface after uploading the tar file: 1.
Administrator’s Guide for SIP-T2xP IP Phones Supporting dedicated bandwidth Improving loss characteristics Avoiding and managing network congestion Shaping network traffic Setting traffic priorities across the network The Best-Effort service is the default QoS model in IP networks. It provides no guarantees for data delivering, which means delay, jitter, packet loss and bandwidth allocation are unpredictable.
Configuring Advanced Features SIP protocol is used for creating, modifying and terminating two-party or multi-party sessions. To ensure good voice quality, SIP packets emanating from IP phones should be configured with a high transmission priority. DSCPs for voice and SIP packets can be specified respectively. Procedure QoS can be configured using the configuration files or locally. Configure the DSCPs for voice Configuration File .cfg packets and SIP packets.
Administrator’s Guide for SIP-T2xP IP Phones Network Address Translation Network Address Translation (NAT) is essentially a translation table that maps public IP address and port combinations to private ones. This reduces the need for a large number of public IP addresses. NAT ensures security since each outgoing or incoming request must first go through a translation process. But in the VoIP environment, NAT breaks end-to-end connectivity.
Configuring Advanced Features 1. Click on Account->Register. 2. Select the desired account from the pull-down list of Account. 3. Select STUN from the pull-down list of NAT. 4. Enter the IP address or the domain name of the STUN server in the STUN Server field. 5. Click Confirm to accept the change. SNMP SNMP (Simple Network Management Protocol) is an Internet-standard protocol for managing devices on IP networks.
Administrator’s Guide for SIP-T2xP IP Phones MIB OID Description (root@localhost) An administratively-assigned name for YEALINK-MIB 1.3.6.1.2.1.37459.2.1.2.0 the IP phone. If the name is unknown, the value is a zero-length string. For example, IPPHONE YEALINK-MIB 1.3.6.1.2.1.37459.2.1.3.0 The physical location of the IP phone. For example, Server Room The time (in milliseconds) since the YEALINK-MIB 1.3.6.1.2.1.37459.2.1.4.0 network management portion of the system was last re-initialized.
Configuring Advanced Features 1. Click on Network->Advanced. 2. In the SNMP block, select the desired value from the pull-down list of Active. 3. Enter the desired port in the Port (1~65535) field. 4. Enter IP address(es) or domain name in the Trusted Address field. Multiple IP addresses are separated by space. 5. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after reboot. 6. Click OK to reboot the IP phone. 802.1X Authentication IEEE 802.
Administrator’s Guide for SIP-T2xP IP Phones IP phones support protocols EAP-MD5, EAP-TLS, PEAP-MSCHAPv2 and EAP-TTLS/EAP-MSCHAPv2 for 802.1X authentication. Procedure 802.1X authentication can be configured using the configuration files or locally. Configure the 802.1X Configuration File .cfg authentication. For more information, refer to 802.1X on page 348. Configure the 802.1X authentication.
Configuring Advanced Features 2. In the 802.1x block, select the desired protocol from the pull-down list of 802.1x Mode. a) If you select EAP-MD5: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. b) If you select EAP-TLS: 1) Enter the user name for authentication in the Identity field. 2) Leave the MD5 Password field blank. 3) In the CA Certificates field, click Browse to select the desired CA certificate (*.pem, *.
Administrator’s Guide for SIP-T2xP IP Phones 5) Click Upload to upload the certificates. c) If you select PEAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 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. 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.
Administrator’s Guide for SIP-T2xP IP Phones 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 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.
Configuring Advanced Features 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 Device Management TR-069 is a technical specification, defined by the Broadband Forum, which defines a mechanism that encompasses secure auto-configuration of a CPE (Customer-Premises Equipment), as well as incorporates other CPE management functions into a common framework.
Administrator’s Guide for SIP-T2xP IP Phones RPC Method Description File types supported by IP phones are: Firmware Image Configuration File This method is used to cause the CPE to upload a specified file to the designated location. File types supported by IP phones are: 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.
Configuring Advanced Features 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.
Administrator’s Guide for SIP-T2xP IP Phones 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 353. 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.
Configuring Advanced Features To configure IPv6 address assignment method via phone user interface: 1. Press Menu->Settings->Advanced Settings (password: admin) ->Network->WAN Port. 2. Press or to select IPv4&IPv6 or IPv6 from the IP Mode field. 3. Press or to highlight IPv6 and press the Enter soft key. 4. Press or to select the desired IPv6 address assignment method. If you select the Static IPv6 Client, configure the IPv6 address and other network parameters in the corresponding fields. 5.
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Configuring Audio Features Configuring Audio Features This chapter provides information for making configuration changes for the following audio features: Headset Prior Dual Headset Audio Codecs Acoustic Clarity Technology Headset Prior Headset prior allows users to use headset preferentially if a headset is physically connected to the IP phone. This feature is especially useful for permanent or full-time headset users.
Administrator’s Guide for SIP-T2xP IP Phones 2. Select the desired value from the pull-down list of Headset Prior. 3. Click Confirm to accept the change. Dual Headset Dual headset allows users to use two headsets on one IP phone. To use this feature, users need to physically connect two headsets to the headset and handset jacks respectively.
Configuring Audio Features 2. Select the desired value from the pull-down list of Dual-Headset. 3. Click Confirm to accept the change. Audio Codecs CODEC is an abbreviation of COmpress-DECompress, capable of coding or decoding a digital data stream or signal by implementing an algorithm. The object of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining the quality.
Administrator’s Guide for SIP-T2xP 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 Configuration Fi
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 357. Configure the ptime. For more information, refer to Audio Codecs on page 357. Configure the codecs to use and adjust the priority of the enabled codecs on a per-line basis.
Administrator’s Guide for SIP-T2xP 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: 200 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 Clarity Technology Acoustic Echo Cancellation Acoustic Echo Cancellation (AEC) is used to remove acoustic echo from a voice communication in order to improve the voice quality. It also increases the capacity achieved through silence suppression by preventing echo from traveling across a network. IP phones employ advanced AEC for hands-free operation. Echo cancellation is achieved using the echo canceller.
Administrator’s Guide for SIP-T2xP IP Phones Voice Activity Detection Voice Activity Detection (VAD) is used in speech processing to detect the presence or absence of human speech. When detecting period of “silence”, VAD replaces that silence efficiently with special packets that indicate silence is occurring. It can facilitate speech processing, and deactivate some processes during non-speech section of an audio session.
Configuring Audio Features Comfort Noise Generation Comfort Noise Generation (CNG) is used to generate background noise for voice communications during periods of silence in a conversation. It is part of the silence suppression or VAD handling for VoIP technology. CNG, in conjunction with VAD algorithms, quickly responds when periods of silence occur and inserts artificial noise until voice activity resumes.
Administrator’s Guide for SIP-T2xP IP Phones Jitter Buffer Jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in even intervals. Jitter is a term indicating variations in packet arrival time, can occur because of network congestion, timing drift or route changes.
Configuring Audio Features 5. Enter the fixed delay time for fixed jitter buffer in the Nominal field. 6. Click Confirm to accept the change.
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Configuring Security Features Configuring Security Features This chapter provides information for making configuration changes for the following security-related features: Transport Layer Security Secure Real-Time Transport Protocol Encrypting Configuration Files Note To use these features correctly, we recommend that IP phones running firmware version 71 or later CANNOT be downgraded to the earlier firmware version.
Administrator’s Guide for SIP-T2xP IP Phones to negotiate the security settings for a network connection using the TLS/SSL network protocol. IP phones supports the following cipher suites for TLS 1.
Configuring Security Features Step1: IP phone sends “Client Hello” message proposing SSL options. Step2: Server responds with “Server Hello” message selecting the SSL options, sends its public key information in “Server Key Exchange” message and concludes its part of the negotiation with “Server Hello Done” message. Step3: IP phone sends session key information (encrypted with server’s public key) in the “Client Key Exchange” message.
Administrator’s Guide for SIP-T2xP IP Phones Configuration changes can be performed using the configuration files or locally. Configure TLS on a per-line .cfg basis. For more information, refer to TLS on page 363. Configure trusted certificates feature. Configure server certificates Configuration File feature. For more information, refer to .cfg TLS on page 363. Upload the trusted certificates. Upload the server certificates.
Configuring Security Features 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. To configure the trusted certificates via web user interface: 1. Click on Security->Trusted Certificates. 2.
Administrator’s Guide for SIP-T2xP IP Phones 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. To configure the server certificates via web user interface: 1. Click on Security->Server Certificates. 2. Select the desired value from the pull-down list of Device Certificates. 3.
Configuring Security Features 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…”. Secure Real-Time Transport Protocol Secure Real-Time Transport Protocol (SRTP) encrypts the RTP streams during VoIP phone calls to avoid interception and eavesdropping.
Administrator’s Guide for SIP-T2xP IP Phones answers the call by responding with a 200 OK message which carries the negotiated RTP encryption algorithm. Example of the RTP encryption algorithm carried in the SDP of the 200 OK message: m=audio 11780 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NGY4OGViMDYzZjQzYTNiOTNkOWRiYzRlMjM0Yzcz a=sendrecv a=ptime:20 a=fmtp:101 0-15 SRTP is configurable on a per-line basis.
Configuring Security Features 4. Select the desired value from the pull-down list of RTP Encryption (SRTP). 5. Click Confirm to accept the change. Encrypting Configuration Files Encrypted configuration files can be downloaded from the provisioning server to protect against unauthorized access and tampering of sensitive information (e.g., login passwords, registration information). Yealink provides configuration encryption tool for encrypting configuration files.
Administrator’s Guide for SIP-T2xP IP Phones For security, administrator should upload encrypted configuration files, .enc and/or .enc files to the root directory of the provisioning server. During auto provisioning, the IP phone requests to download .cfg file first. If the downloaded configuration file is encrypted, the phone will request to download .enc file (if enabled) and decrypt .
Configuring Security Features 5. Click Encrypt to encrypt the configuration file(s). 6. Click OK. The target directory will be automatically opened. You can find the encrypted CFG file(s), encrypted key file(s) and an Aeskey.txt file storing plaintext AES key(s). Procedure Encryption method can be configured using the configuration files. Configure the encryption method. Configuration File .cfg Configure AES keys.
Administrator’s Guide for SIP-T2xP IP Phones To configure AES keys via web user interface: 1. Click on Settings->Auto Provision. 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. 218 Click Confirm to accept the change.
Upgrading Firmware 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-T20P 9.x.x.x.
Administrator’s Guide for SIP-T2xP IP Phones A dialog box pops up to prompt “Firmware of the SIP Phone will be updated. It will take 5 minutes to complete. Please don't power off!”. 5. Note Click OK to confirm the upgrading. Do not unplug the network and power cables when the IP phone is upgrading firmware. Do not close the browser when the IP phone is upgrading firmware via web user interface.
Upgrading Firmware Upgrading Firmware on page 368. 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 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 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 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. When you leave it blank or enter 0, this dial-now rule will apply to all lines. 3.
Administrator’s Guide for SIP-T2xP IP Phones end of the default soft key list, the default soft keys are displayed on the LCD screen by default. Procedure Use the following procedures to customize a softkey layout template. To customize a softkey layout template: 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 Local Contact File You can add contacts one by one on the IP phone directly. You can also add multiple contacts at a time and/or share contacts between IP phones using the local contact template file. After setup, place the template file to the provisioning server and specify the access URL of the template file in the configuration files.
Administrator’s Guide for SIP-T2xP IP Phones mobile_number="" specifies the mobile number of the contact. other_number="" specifies the other number of the contact. line="" specifies the line you want to add this contact to. ring="" specifies the ring tone for this contact. 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 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 IP Phones the dial-now rule template. For more information, refer to Access URL of Dial-now Template on page 371. Configure the access URL of the softkey layout template. Configuration File .cfg For more information, refer to Access URL of Softkey Layout Template on page 371. Configure the access URL of the local contact file. Configuration File .cfg For more information, refer to Access URL of Local Contact File on page 374.
Troubleshooting Troubleshooting This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using SIP-T2xP IP phones. Troubleshooting Methods IP phones can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which can help an administrator more easily find the system problem and fix it.
Administrator’s Guide for SIP-T2xP 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 reboot. 4. Click OK to reboot the IP phone. After reboot, the system log level is set as 6, the administrator debug level. Note Administrator level debugging may make some sensitive information become accessable (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 reboot. 5. Click OK to reboot the IP phone. The system log will be exported successfully to the desired syslog server after 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 IP Phones The following figure shows a portion of a log file: Capturing Packets You can capture packet in two ways: capturing the packet via web user interface or using the Ethernet software. You can analyze the packet captured for troubleshooting purpose. To capture packet via web user interface: 234 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 end capturing.
Troubleshooting 5. Click Export to open the file download window, and then save the file to your local system. To capture packet using the Ethernet software: Connect the Internet port of the IP phone and the PC to the same HUB, and then use Sniffer, Ethereal or Wireshark software to capture the signal traffic.
Administrator’s Guide for SIP-T2xP IP Phones 2. Select the desired value from the pull-down list of Watch Dog. 3. Click Confirm to accept the change. Getting Information from Status Indicators Status indicators may consist of the power LED, MESSAGE key LED, line key indicator, headset key indicator and the on-screen icon or error messages.
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. Troubleshooting Solutions This section describes solutions to common issues that may occur while using the IP phone. Upon encountering a scenario not listed in this section, contact your Yealink reseller for further support.
Administrator’s Guide for SIP-T2xP IP Phones Why does the IP phone display “No Service”? The LCD screen prompts “No Service” message when there is no available SIP account on the IP phone. Do one of the following: Ensure that an account is actively registered on the IP phone at the path Menu->Status->More->Accounts. Ensure that the SIP account parameters have been set up correctly.
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 IP Phones Why does the IP phone use DOB format logo file instead of popular BMP, JPG and so on? 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 less space. Tools for converting BMP format to DOB format are available. For more information, refer to Yealink SIP-T2 Series/T3 Series/VP530 IP Phones Auto Provisioning Guide.
Troubleshooting Why doesn’t the IP phone update the configuration? Do one of the following: Ensure that the configuration is set correctly. Reboot the IP phone. Some configurations require a reboot to take effect. Ensure that the configuration is applicable to the IP phone model. The configuration may depend on support from the server. What do “on code” and “off code” mean? They are codes that the IP phone sends to the server when a certain action takes place.
Administrator’s Guide for SIP-T2xP 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.
Troubleshooting Phone Model LCD Logo Display Line Key Memory Key XML SMS and an icon messages line) via web user Browser interface) 243
Administrator’s Guide for SIP-T2xP IP Phones 244
Appendix Appendix Appendix A: Glossary 802.1x--an IEEE Standard for port-based Network Access Control (PNAC). It is part of the IEEE 802.1 group of networking protocols. It provides an authentication mechanism to devices wishing to attach to a LAN or WLAN. ACD (Automatic Call Distribution)--used to distribute calls from large volumes of incoming calls to the registered IP phone users. ACS (Auto Configuration server)--responsible for auto-configuration of the Central Processing Element (CPE).
Administrator’s Guide for SIP-T2xP IP Phones HTTPS (Hypertext Transfer Protocol over Secure Socket Layer)--a widely-used communications protocol for secure communication over a network. IEEE (Institute of Electrical and Electronics Engineers)--a non-profit professional association headquartered in New York City that is dedicated to advancing technological innovation and excellence.
Appendix Appendix B: Time Zones Time Zone Time Zone Name −11:00 Samoa −10:00 United States-Hawaii-Aleutian −10:00 United States-Alaska-Aleutian −09:00 United States-Alaska Time −08:00 Canada(Vancouver, Whitehorse) −08:00 Mexico(Tijuana, Mexicali) −08:00 United States-Pacific Time −07:00 Canada(Edmonton, Calgary) −07:00 Mexico(Mazatlan, Chihuahua) −07:00 United States-Mountain Time −07:00 United States-MST no DST −06:00 Canada-Manitoba(Winnipeg) −06:00 Chile(Easter Islands) −06:0
Administrator’s Guide for SIP-T2xP IP Phones Time Zone 248 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 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 Netherlands(Amsterdam) +01:00 Namibia
Appendix Time Zone Time Zone Name +05:00 Kazakhstan(Aqtobe) +05:00 Kyrgyzstan(Bishkek) +05:00 Pakistan(Islamabad) +05:00 Russia(Chelyabinsk) +05:30 India(Calcutta) +06:00 Kazakhstan(Astana, Almaty) +06:00 Russia(Novosibirsk, Omsk) +07:00 Russia(Krasnoyarsk) +07:00 Thailand(Bangkok) +08:00 China(Beijing) +08:00 Singapore(Singapore) +08:00 Australia(Perth) +09:00 Korea(Seoul) +09:00 Japan(Tokyo) +09:30 Australia(Adelaide) +09:30 Australia(Darwin) +10:00 Australia(Sydney, Me
Administrator’s Guide for SIP-T2xP IP Phones Appendix C: Configuration Parameters This appendix describes configuration parameters in the configuration files for each feature. The configuration files are .cfg and .cfg. Setting Parameters in Configuration Files You can set parameters in the configuration files to configure IP phones. The .cfg and .cfg files are stored on the provisioning server.
Appendix Static Network Settings 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.internet_port.type = 2 Parameter- Configuration File network.ip_address_mode .cfg Configures the IP address mode.
Administrator’s Guide for SIP-T2xP IP Phones port type is configured as Static IP Address and the IP address mode is configured as IPv4 or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank Range Not Applicable Example network.internet_port.ip = 192.168.1.20 Parameter- Configuration File network.internet_port.mask .
Appendix Range Example Not Applicable network.internet_port.gateway = 192.168.1.254 Parameter- Configuration File network.primary_dns .cfg Configures the primary DNS server when the Internet port type is configured as Static IP Address and the IP address mode is configured Description as IPv4 or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank Range Not Applicable Example network.
Administrator’s Guide for SIP-T2xP IP Phones PPPoE 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.internet_port.type= 1 Parameter- Configuration File network.pppoe.user .
Appendix Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format String Default Value Blank Range Not Applicable Example network.pppoe.password = yealink123 Internet and PC Ports Transmission Methods Internet Port Transmission Method Parameter- Configuration File network.internet_port.speed_d .cfg uplex Specifies the transmission method of Internet Description port. Note: We recommend that you do not change this parameter.
Administrator’s Guide for SIP-T2xP IP Phones Default Value 0 Valid values are: 0-Auto negotiate Range 1-Full duplex, 10Mbps 2-Full duplex, 100Mbps 3-Half duplex, 10Mbps 4-Half duplex, 100Mbps Example network.pc_port.speed_duplex = 0 PC Port Mode Parameter- Configuration File network.PC_port.enable .cfg Enables or disables the PC port. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Appendix Parameter- Configuration File network.pc_port.ip .cfg Configures the IP address for the PC port when the PC port is configured as Router. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value 10.0.0.1 Range Not Applicable Example network.pc_port.ip = 10.0.0.1 Parameter- Configuration File network.pc_port.mask .
Administrator’s Guide for SIP-T2xP IP Phones Example network.pc_port.dhcp_server = 1 Parameter- Configuration File network.dhcp.start_ip .cfg Configures the start IP address that the IP phone assigns for the PC attached to the PC Description 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. Format IP Address Default Value 10.0.0.10 Range Not Applicable Example network.dhcp.start_ip = 10.
Appendix Replaced, Line ID Enabled/Disabled: Enables or disables the replace rule. Prefix: Specifies the string you want to replace. Replaced: Specifies the alternate string instead of what the user enters. Line ID: Specifies the desired line to apply this replace rule. The digit 0 stands for all lines. X ranges from 1 to 100. Note: Multiple line IDs are separated by comma.
Administrator’s Guide for SIP-T2xP IP Phones comma. Format String, Integer Default Value Blank Dial-now Rules: Not Applicable Valid values of Line ID are: Range 0 to 6 (for T28P) 0 to 3 (for T26P/T20P) 0 to 2 (for T20P) Example dialnow.item.1 = 2216,1,2,3 Parameter- Configuration File phone_setting.dialnow_delay .cfg Configures the delay time (in seconds) for the dial-now rule.
Appendix 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 Description entered numbers. Note: The value must be larger than the minimum length.
Administrator’s Guide for SIP-T2xP IP Phones 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 Not Applicable 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.
Appendix Default Value 6 Range 1 to 10 Example phone_setting.contrast = 6 Backlight Parameter- Configuration File phone_setting.active_backlight .cfg _level Configures the backlight idle intensity used to adjust the backlight intensity of the LCD Description screen Level 3 is the brightest. Note: It is only applicable to the SIP-T28P IP phone. Format Integer Default Value 2 Range 1 to 3 Example phone_setting.
Administrator’s Guide for SIP-T2xP IP Phones Example phone_setting.backlight_time = 30 User Password Parameter- Configuration File security.user_password .cfg Configures a new user password for the IP phone. Description The IP phone uses “user” as the default user password. Note: IP phones support ASCII characters 32-126(0x20-0x7E) only in passwords. Format username:new password Default Value user Range ASCII characters 32-126(0x20-0x7E) Example security.
Appendix phone_setting.lock .cfg Configures the type of phone lock. Menu Key: The Menu soft key and MESSAGE key are locked (For T20P, the MENU key is locked). Function Keys: MESSAGE, RD, CONF, HOLD, MUTE, TRAN, OK, X, navigation keys, soft keys, line keys and memory keys are locked (For T22P, CONF, HOLD, MUTE and memory keys do not exist; For T20P, the MUTE key, soft keys and memory keys do not exist, but the additional MENU and Directory keys are locked).
Administrator’s Guide for SIP-T2xP IP Phones IP phone is locked, you can use the default password “123” to unlock it. Format Not Applicable Default Value 123 Range 0 to 15 characters Example phone_setting.phone_lock.unlock_pin = 123 Parameter- Configuration File phone_setting.phone_lock.loc .cfg k_time_out Configures the IP phone to automatically lock the keypad after a delay time (in seconds). If set to 0 (0s), the keypad will not be locked Description automatically.
Appendix 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.org Range Not Applicable Example local_time.ntp_server2 = cn.pool.ntp.org Parameter- Configuration File local_time.
Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File local_time.time_zone_name .cfg Configures the desired time zone name. Description For more available time zone name list, refer to Appendix B: Time Zones on page 247. Format String Default Value China(Beijing) Range Not Applicable Example local_time.time_zone_name = China(Beijing) DST Parameter- Configuration File local_time.summer_time .
Appendix 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.
Administrator’s Guide for SIP-T2xP IP Phones If “local_time.dst_time_type” is set to 1 (By Week), use the mapping: Month: 1=Jan, 2=Feb,…, 12=Dec Week of Month: 1=the first week in a month,…, 5=the last week in a month Day of Week: 1=Mon, 2=Tues,…, 7=Sun Hour of Day: 0=1am, 1=2am,…, 23=12pm Note: It works only if the parameter “local_time.summer_time” is set to 1 (Enabled).
Appendix hour format. Format Integer Default Value 1 Range Example 0-12 Hour 1-24 Hour local_time.time_format = 1 Date Format Parameter- Configuration File local_time.date_format .cfg Configures the date format. Description IP phones support various date formats. You can change the desired format according to your requirement.
Administrator’s Guide for SIP-T2xP IP Phones Note: The language packs you load are dependent on available language packs from the provisioning server. You can download the language pack to the phone user interface only. Format URL Default Value Blank Range Not Applicable The following example uses HTTP to download the language pack Example “lang+English.txt”(English) from the provisioning server 192.168.10.25. gui_lang.url = http://192.168.10.25/lang+English.
Appendix user interface depends on the language preferences of your browser. If the language of your browser is not supported by the IP phone, the web user interface will use English by default. Format String Default Value Not Applicable Valid values are: English Deutsch Range French Italian Portuguese Spanish Turkish Example lang.wui = English Logo Customization Parameter- Configuration File phone_setting.lcd_logo.mode .cfg Configures the logo mode of the LCD screen.
Administrator’s Guide for SIP-T2xP IP Phones value is 1. Valid values are: 0-Disabled 1-System logo Range 2-Custom logo Note: For the SIP-T28 IP phone, valid values are 1(System logo) and 2(Custom logo). For the SIP-T20P IP phones, valid values are 0(Disabled) and 1(Enabled). 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 phone.
Appendix Key as Send Parameter- Configuration File features.pound_key.mode .cfg Configures the "#" or "*" key as the send key. If set to 0 (Disabled), neither “#” nor “*” can be used as a send key. Description If set to 1(# key), the pound key is used as the send key. If 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.pound_key.
Administrator’s Guide for SIP-T2xP IP Phones Hotline Parameter- Configuration File features.hotline_number .cfg Configures the hotline number. It specifies 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. Format String Default Value Blank Range Not Applicable Example features.hotline_number = 3601 Parameter- Configuration File features.
Appendix Call Log Parameter- Configuration File features.history_save_display .cfg Enables or disables the IP phone to display the Save Call Log option on the web user interface. Description If set to 0 (Disabled), the Save Call Log option is hidden on the web user interface. If set to 1 (Enabled), you can enable or disable call log feature via web user interface. Format Boolean Default Value 1 Range 0-Disabled 1-Enabled Example features.
Administrator’s Guide for SIP-T2xP IP Phones displaying on the LCD screen, the IP phone does not log the missed call in the Missed Calls list. If set to 1 (Enabled), a prompt message " New Missed Call(s)" along with an indicator icon is displayed on the IP phone idle screen when the IP phone misses calls. X ranges from 1 to 6. Format Boolean Default Value 1 Range Example 0-Disabled 1-Enabled account.1.missed_calllog = 1 Live Dialpad Parameter- Configuration File phone_setting.
Appendix Default Value 4 Range 1 to 14 Example phone_setting.inter_digit_time = 1 Call Waiting Parameter- Configuration File call_waiting.enable .cfg Enables or disables call waiting feature. If set to 0 (Disabled), a new incoming call is Description automatically rejected by the IP phone with a busy message while during a call. If set to 1 (Enabled), the LCD screen presents a new incoming call while during a call.
Administrator’s Guide for SIP-T2xP IP Phones Auto Redial Parameter- Configuration File auto_redial.enable .cfg Enables or disables the IP phone to automatically redial the called number when Description it is busy. If set to 1 (Enabled), the IP phone dials the previous dialed out number automatically when the dialed number is busy. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example auto_redial.enable = 1 Parameter- Configuration File auto_redial.
Appendix Range 1 to 300 Example auto_redial.times = 10 Auto Answer Parameter- Configuration File account.x.auto_answer .cfg Enables or disables auto answer feature for account x. If set to 1 (Enabled), the IP phone can Description automatically answer an incoming call. X ranges from 1 to 6. Note: The IP phone cannot automatically answer the incoming call during a call even if auto answer is enabled. Format Boolean Default Value 0 Range Example 0-Disabled 1-Enabled account.1.
Administrator’s Guide for SIP-T2xP IP Phones 1-Enabled Example features.call_completion_enable = 1 Anonymous Call Parameter- Configuration File account.x.anonymous_call .cfg Enables or disables anonymous call feature for account x. If set to 1 (Enabled), the IP phone blocks its Description 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.
Appendix account.x.anonymous_call_on .cfg code Configures the anonymous call on code to activate the server-side anonymous call feature for account x (optional). Description X ranges from 1 to 6. Note: It works only if the parameter “account.x.send_anonymous_code” is set to 1 (Enabled). Format String Default Value Blank Range Not Applicable Example account.1.anonymous_call_oncode = *72 Parameter- Configuration File account.x.anonymous_call_off .
Administrator’s Guide for SIP-T2xP IP Phones automatically rejects incoming calls from users enabled anonymous call feature. The anonymous user’s phone LCD screen presents “Anonymity Disallowed”. X ranges from 1 to 6. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example account.1.reject_anonymous_call = 1 Parameter- Configuration File account.x.anonymous_reject_ .
Appendix Do Not Disturb Return Message When DND Parameter- Configuration File features.dnd_refuse_code .cfg Configures return codes and reason of the SIP response message when rejecting an incoming call for DND. A specific reason is Description displayed on the caller’s phone LCD screen. If set to 486 (Busy here), the caller’s phone LCD screen displays the reason “Busy here” when the callee enables DND feature.
Administrator’s Guide for SIP-T2xP IP Phones DND in Phone Mode Parameter- Configuration File features.dnd.enable .cfg Enables or disables DND feature. Description If set to 1 (Enabled), the IP phone rejects incoming calls on all accounts. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example features.dnd.enable = 1 Parameter- Configuration File features.dnd.on_code .
Appendix If set to 1 (Enabled), the IP phone rejects incoming calls on account x. X ranges from 1 to 6. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example account.1.dnd.enable = 1 Parameter- Configuration File account.x.dnd.on_code .cfg Configures the DND on code to activate the Description server-side DND feature for account x (optional). X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example account.1.dnd.
Administrator’s Guide for SIP-T2xP IP Phones Busy Tone Delay 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. Description When one party releases the call, a busy tone is audible to the other party indicating that the call connection breaks. If set to 3 (3s), a busy tone is audible for 3 seconds on the IP phone.
Appendix 486-Busy here Example features.normal_refuse_code = 486 180 Ring Workaround 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 set to 1 (Enabled), the IP phone resumes and plays the local ringback tone upon a subsequent 180 message received. Format Boolean Default Value 1 Range Example 0-Disabled 1-Enabled phone_setting.
Administrator’s Guide for SIP-T2xP IP Phones SIP Session Timer 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 = 0.5 Parameter- Configuration File account.x.advanced.timer_t2 .
Appendix to clear messages between the SIP Client and SIP Server. X ranges from 1 to 6. Format Float Default Value 5 Range 2.5 to 60 Example account.1.advanced.timer_t4 = 5 Session Timer Parameter- Configuration File account.x.session_timer.enable .cfg Enables or disables the session timer for account x. Description If set to 1 (Enabled), IP phone sends periodic re-INVITE requests to refresh the session during a call. X ranges from 1 to 6.
Administrator’s Guide for SIP-T2xP IP Phones 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 set to 0 (UAC), refreshing the session is Description performed by the IP phone. If set to 1 (UAS), refreshing the session is performed by a SIP server. X ranges from 1 to 6. Format Integer Default Value 0 Valid values are: Range 0-UAC 1-UAS Example account.1.
Appendix tone every 30 seconds when there is a hold call on the IP phone. Note: It works only if the parameter “features.play_hold_tone.enable” is set to 1 (Enabled). Format Integer Default Value 30 Range Not Applicable Example features.play_hold_tone.delay = 30 Parameter- Configuration File sip.rfc2543_hold .cfg Configures whether RFC 2543 (c=0.0.0.0) outgoing hold signaling is used.
Administrator’s Guide for SIP-T2xP IP Phones forward feature for each account. Format Integer Default Value 0 Range Example 0-Phone 1-Custom features.fwd_mode = 0 Call Forward in Phone Mode Always Forward Parameter- Configuration File forward.always.enable < y0000000000xx >.cfg Enables or disables always forward feature. Description If set to 1 (Enabled), incoming call are forwarded to the destination number immediately.
Appendix Format String Default Value Blank Range Not Applicable Example forward.always.on_code = *72 Parameter- Configuration File forward.always.off_code < y0000000000xx >.cfg Configures the always forward off code to Description deactivate the server-side always forward feature. Format String Default Value Blank Range Not Applicable Example forward.always.off_code = *73 Busy Forward Parameter- Configuration File forward.busy.enable < y0000000000xx >.
Administrator’s Guide for SIP-T2xP IP Phones Range Not Applicable Example forward.busy.target = 3602 Parameter- Configuration File forward.busy.on_code < y0000000000xx >.cfg Configures the busy forward on code to Description activate the server-side busy forward feature. Format String Default Value Blank Range Not Applicable Example forward.busy.on_code = *74 Parameter- Configuration File forward.busy.off_code < y0000000000xx >.
Appendix Example forward.no_answer.enable = 1 Parameter- Configuration File forward.no_answer.target < y0000000000xx >.cfg Description Configures the destination number of the no answer forward. Format String Default Value Blank Range Not Applicable Example forward.no_answer.target = 3603 Parameter- Configuration File forward.no_answer.timeout < y0000000000xx >.cfg Configures a period of ring time to wait Description before forwarding the incoming call.
Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File forward.no_answer.off_code < y0000000000xx >.cfg Configures the no answer forward off code Description to deactivate the server-side no answer forward feature. Format String Default Value Blank Range Not Applicable Example forward.no_answer.off_code = *77 Call Forward in Custom Mode Always Forward Parameter- Configuration File account.x.always_fwd.enable .
Appendix Parameter- Configuration File account.x.always_fwd.on_code .cfg Configures the always forward on code Description activate the server-side always forward feature for account x. X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example account.1.always_fwd.on_code = *72 Parameter- Configuration File account.x.always_fwd.off_code .
Administrator’s Guide for SIP-T2xP IP Phones Example account.1.busy_fwd.enable = 1 Parameter- Configuration File account.x.busy_fwd.target .cfg Configures the destination number of the Description busy forward for account x. X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example account.1.busy_fwd.target = 3602 Parameter- Configuration File account.x.busy_fwd.on_code .
Appendix No Answer Forward Parameter- Configuration File account.x.timeout_fwd.enable .cfg Enables or disables no answer forward feature for account x. Description If set to 1 (Enabled), incoming calls to the account x are forward to the destination number after a period of ring time. X ranges from 1 to 6. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example account.1.timeout_fwd.enable = 1 Parameter- Configuration File account.x.timeout_fwd.target .
Administrator’s Guide for SIP-T2xP IP Phones Range 0 to 20 Example account.1.timeout_fwd.timeout = 2 Parameter- Configuration File account.x.timeout_fwd.on_code .cfg Configures the no answer forward on 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 Not Applicable Example account.1.timeout_fwd.on_code = *76 Parameter- Configuration File account.x.timeout_fwd.off_code .
Appendix Example forward.international.enable = 1 Call Transfer Parameter- Configuration File transfer.blind_tran_on_hook_ena .cfg ble Description Enables or disables the IP phone to complete the blind transfer through on-hook. Format Boolean Default Value 1 Range 0-Disabled 1-Enabled Example transfer.blind_tran_on_hook_enable = 1 Parameter- Configuration File transfer.on_hook_trans_enable .
Administrator’s Guide for SIP-T2xP IP Phones Example transfer.semi_attend_tran_enable = 1 Network Conference Parameter- Configuration File account.x.conf_type .cfg Configures the conference type for account x. If set to 0 (Local Conference), conferences Description are set up on the IP phone locally. If set to 2 (Network Conference), conferences are set up by the server. X ranges from 1 to 6.
Appendix Transfer on Conference Hang Up Parameter- Configuration File transfer.tran_others_after_conf_e .cfg nable Enables or disables Transfer on Conference Hang Up feature. If enabled, the other two parties remain Description connected when the conference initiator drops the conference call. Note: It is only applicable to the local conference. Format Boolean Default Value 0 Range Example 0-Disabled 1-Enabled transfer.
Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File features.pickup.direct_pickup_c .cfg ode Configures the directed call pickup code on a phone basis. Description Note: The directed call pickup code configured on a per-line basis takes precedence over that configured on a phone basis. Format String Default Value Blank Range Not Applicable Example features.pickup.direct_pickup_code = *97 Per-line Basis Parameter- Configuration File account.x.
Appendix the GPickup soft key when the IP phone is off-hook. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example features.pickup.group_pickup_enable = 1 Parameter- Configuration File features.pickup.group_pickup_c .cfg ode 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.
Administrator’s Guide for SIP-T2xP IP Phones Dialog-Info Call Pickup Parameter- Configuration File account.x.dialoginfo_callpickup .cfg Configures Dialog-Info Call Pickup feature for account x. Description If set to 1 (Enabled), call pickup is implemented through SIP signals. X ranges from 1 to 6. Format Boolean Default Value 0 Range Example 0-Disabled 1-Enabled account.1.dialoginfo_callpickup = 1 Web Server Type Parameter- Configuration File wui.http_enable .
Appendix phone will reboot to make the change take effect. Format Integer Default Value 80 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 Range 0-Disabled 1-Enabled Example wui.
Administrator’s Guide for SIP-T2xP IP Phones Calling Line Identification Presentation Parameter- Configuration File account.x.cid_source .cfg Configures the presentation of the caller identity for account x. 0-FROM (Derives the name and number of the caller from the “From” header). 1-PAI (Derives the name and number of the caller from the “PAI” header. If the server does not send the “PAI” header, displays “anonymity” on the callee’s phone).
Appendix 2-RFC 4916 (Derives the name and number of the callee from “From” header in the Update message). When the RFC 4916 is enabled on the IP phone, the caller sends the SIP request message which contains the from-change tag in the Supported header. The caller then receives an UPDATE message from the callee, and displays the identity in the From header. 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.
Administrator’s Guide for SIP-T2xP IP Phones 2-SIP INFO 3-AUTO or SIP INFO Example account.1.dtmf.type = 1 Parameter- Configuration File account.x.dtmf.dtmf_payload .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 .
Appendix Suppress DTMF Display Parameter- Configuration File features.dtmf.hide .cfg Enables or disables the IP phone to suppress Description the display of DTMF digits. If set to 1 (Enabled), the DTMF digits are displayed as asterisks. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example features.dtmf.hide = 1 Parameter- Configuration File features.dtmf.hide_delay .
Administrator’s Guide for SIP-T2xP IP Phones the specified DTMF digits to the server for completing call transfer when pressing the transfer key during a call. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example features.dtmf.replace_tran = 1 Parameter- Configuration File features.dtmf.transfer .cfg Configures the DTMF digits to be transmitted to complete the transfer. Description Note: It works only if the parameter “features.dtmf.
Appendix 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 set to 0 (Disabled), the microphone is un-muted for incoming calls. If set to 1 (Enabled), the microphone is muted for intercom calls. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example features.intercom.
Administrator’s Guide for SIP-T2xP IP Phones 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. If set to 0 (Disabled), the IP phone handles an Description incoming intercom call like a waiting call while there is already an active call on the IP phone.
Appendix X ranges from 1 to 6. Format Boolean Default Value 1 Range 0-Disabled 1-Enabled Example account.1.alert_info_url_enable = 1 Parameter- Configuration File distinctive_ring_tones.alert_info.x .cfg .text Configures the texts to map the keywords Description contained in the SIP header. X ranges from 1 to 10. Format String Default Value Blank Range Not Applicable Example distinctive_ring_tones.alert_info.1.
Administrator’s Guide for SIP-T2xP IP Phones Tones Parameter- Configuration File voice.tone.country .cfg Description Configures the country tone for the IP phone.
Appendix voice.tone.ring .cfg voice.tone.busy voice.tone.congestion voice.tone.callwaiting voice.tone.dialrecall voice.tone.info voice.tone.stutter voice.tone.message voice.tone.autoanswer Configures the tone for each condition. tonelist = element[,element] [,element]… Where element = [!]freq1[+freq2][+freq3][+freq4] /duration Freq: the frequency of the tone (ranges from 200 to 7000 Hz). If set to 0 (0Hz), it means the tone is not played.
Administrator’s Guide for SIP-T2xP IP Phones remote_phonebook.data.x.url .cfg Configures the access URL of the remote XML phone book. Description X ranges from 1 to 5. Format URL Default Value Blank Range Not Applicable remote_phonebook.data.1.url = Example http://192.168.1.20/phonebook.xml Parameter- Configuration File remote_phonebook.data.x.nam .cfg e Description Configures the name of the remote phone book.
Appendix Configures how often to refresh the local cache of the remote phone book. Description If set to 3600 (3600s), the IP phone refreshes the local cache of the remote phone book every 3600 seconds. Format Integer Default Value 21600 Range 120 to 2592000 Example features.remote_phonebook.flash_time = 1800 LDAP Parameter- Configuration File ldap.name_filter .cfg Configures the name attribute for LDAP searching.
Administrator’s Guide for SIP-T2xP IP Phones Format String Default Value Blank Range Not Applicable ldap.number_filter = (|(telephoneNumber=%)(Mobile=%)(ipPh one=%)) Example When the number prefix of the telephoneNumber, Mobile or ipPhone of the contact record matches the search criteria, the record will be displayed on the LCD screen. Parameter- Configuration File ldap.host .cfg Description 322 Configures the domain name or IP address of the LDAP server.
Appendix Parameter- Configuration File ldap.base .cfg Configures the LDAP search base which corresponds to the location in the LDAP Description phone book from which the LDAP search request begins. The search base narrows the search scope and decreases directory search time. Format String Default Value Blank Range Not Applicable Example ldap.base = dc=yealink,dc=cn Parameter- Configuration File ldap.user .
Administrator’s Guide for SIP-T2xP IP Phones Default Value Blank Range Not Applicable Example ldap.password = secret Parameter- Configuration File ldap.max_hits .cfg Configures the maximum number of search results to be returned by the LDAP server. If the value of the “Max.Hits” is blank, the LDAP Description server will return all searched results. Please note that a very large value of the “Max.
Appendix configure multiple number attributes separated by space. Format String Default Value Blank Range Not Applicable Example ldap.numb_attr = telephoneNumber Parameter- Configuration File ldap.display_name .cfg Configures the display name of the contact Description record displayed on the LCD screen. Note: It must start with “%” symbol. Format String Default Value Blank Range Not Applicable ldap.
Administrator’s Guide for SIP-T2xP IP Phones call. Format Boolean Default Value 0 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 Description search results in alphabetical order or numerical order. Format Boolean Default Value 0 Range Example 0-Disabled 1-Enabled ldap.ldap_sort = 1 BLF Visual and Audio Alert for BLF Pickup Parameter- Configuration File features.
Appendix Parameter- Configuration File features.pickup.blf_audio_enable .cfg Enables or disables the IP phone to play an Description alert tone when the monitored user receives an incoming call. Format Boolean Default Value 0 Range Example 0-Disabled 1-Enabled features.pickup.blf_audio_enable = 1 LED Off in Idle Parameter- Configuration File features.blf_and_callpark_idle_le .cfg d_enable Description Enables or disabled LED off in idle feature.
Administrator’s Guide for SIP-T2xP IP Phones Range Not Applicable Example account.1.music_server_uri =<10.1.3.165> Parameter- Configuration File account.x.acd.enable .cfg ACD Enables or disables ACD feature for account Description x. X ranges from 1 to 6. Format Boolean Default Value 0 Value 0-Disabled 1-Enabled Example account.1.acd.enable = 1 Parameter- Configuration File account.x.acd.available MAC.
Appendix Format Boolean Default Value 0 Value 0-Disabled 1-Enabled Example acd.auto_available = 1 Parameter- Configuration File acd.auto_available_timer .cfg Configures the length of time (in seconds) before the IP phone state is automatically Description changed to available. Note: It works only if the parameter “acd.auto_available” is set to 1 (Enabled). Format Integer Default Value 60 Value 0 to 120 Example acd.
Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File account.x.subscribe_mwi_expires .cfg Configures MWI subscribe expiry time (in seconds) for account x. The IP phone is able to successfully refresh the SUBCRIBE for message-summary events Description before expiration of the SUBSCRIBE dialog. X ranges from 1 to 6. Note: It works only if the parameter “account.x.subscribe_mwi” is set to 1 (Enabled). Format Integer Default Value 3600 Value 0 to 84600 Example account.
Appendix Format Boolean Default Value 0 Value Example 0-Disabled 1-Enabled account.1.subscribe_mwi_to_vm = 0 Sending RTP Stream Parameter- Configuration File multicast.codec .cfg Description Configures a multicast codec for the IP phone to use to send an RTP stream. Format string Default Value G722 Valid values are: Range Example PCMU PCMA G729 G722 G726-16 G726-24 G726-32 G726-40 G723_53 multicast.
Administrator’s Guide for SIP-T2xP IP Phones Format Boolean Default Value 1 Range 0-Disabled 1-Enabled Example multicast.receive_priority.enable =1 Parameter- Configuration File multicast.receive_priority.priority < y0000000000xx >.cfg Configures the priority of multicast paging calls. Description 1 is the highest priority, 10 is the lowest priority. If set to 0, all incoming multicast paging calls will be automatically ignored.
Appendix number that the IP phone listens to. X ranges from 1 to 10. Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255. Format String Default Value Blank Range Not Applicable Example multicast.listen_address.1.ip_address = 224.5.6.20:10008 Action URL Parameter- Configuration File action_url.setup_completed .cfg action_url.log_on action_url.log_off action_url.register_failed action_url.off_hook action_url.on_hook action_url.incoming_call action_url.
Administrator’s Guide for SIP-T2xP IP Phones action_url.unmute action_url.missed_call action_url.call_terminated action_url.busy_to_idle action_url.idle_to_busy action_url.ip_change action_url.forward_incoming_call action_url.reject_incoming_call action_url.answer_new_incoming_ call action_url.transfer_finished action_url.transfer_failed Configures the URL for the predefined event. The value format is: http(s)://IP address of server/help.xml? variable name=variable value.
Appendix Action URI Parameter- Configuration File features.action_uri_limit_ip .cfg Configures the address(es) from which Action URI will be accepted. For discontinuous IP addresses, each IP address is separated by comma. For continuous IP addresses, the format likes *.*.*.* and the “*” stands for the values 0~255. Description For example: 10.10.*.* stands for the IP addresses that range from 10.10.0.0 to 10.10.255.255.
Administrator’s Guide for SIP-T2xP IP Phones yealink.pbx.com Parameter- Configuration File account.x.sip_server.y.port .cfg Configures the port of the SIP server for Description account x. X ranges from 1 to 6. Y ranges from 1 to 2. Format Integer Default Value 5060 Range 0 to 65535 Example account.1.sip_server.1.port = 5060 Parameter- Configuration File account.x.sip_server.y.expires .
Appendix Example account.1.sip_server.1.retry_counts = 3 Fallback Mode Parameter- Configuration File account.x.fallback.redundancy_ty .cfg pe Configures the registration mode for the IP Description phone in fallback mode. X ranges from 1 to 6. Format Integer Default Value 0 Valid values are: Range 0-Concurrent registration 1-Successive registration Example account.1.fallback.redundancy_type = 0 Parameter- Configuration File account.x.fallback.timeout .
Administrator’s Guide for SIP-T2xP IP Phones Configures the way in which the phone fails back to the primary server for call Description control in the failover mode. X ranges from 1 to 6. Y ranges from 1 to 2. Format Integer Default Value 0 Valid values are: 0-newRequests: all requests are sent to the primary server first, regardless of the last server that was used.
Appendix X ranges from 1 to 6. Y ranges from 1 to 2. Format Integer Default Value 3600 Range 0, 60 to 65535 Example account.1.sip_server.1.failback_timeout = 3600 Parameter- Configuration File account.x.sip_server.y.register_on_ .cfg enable Enables or disables the IP phone to register to the secondary server before sending Description requests to the secondary server in the failover mode. X ranges from 1 to 6. Y ranges from 1 to 2.
Administrator’s Guide for SIP-T2xP IP Phones Default Value 0 Valid values are: 0-UDP Range 1-TCP 2-TLS 3-DNS-NAPTR Example account.1.transport = 3 Parameter- Configuration File account.x.naptr_build .cfg Configures UDP SRV query or TCP/TLS SRV query for the IP phone to be performed Description when no result is returned from NAPTR 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.
Appendix Parameter- Configuration File network.lldp.packet_interval .cfg Configures the amount of time (in seconds) between the transmissions of LLDP packet. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. It works only if the parameter “network.lldp.enable” is set to 1 (Enabled). Format Integer Default Value 60 Range 1 to 3600 Example network.lldp.
Administrator’s Guide for SIP-T2xP IP Phones Format Integer Default Value 1 Range 1 to 4094 Example network.vlan.internet_port_vid = 1 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.
Appendix Parameter- Configuration File network.vlan.pc_port_vid .cfg Configures the VLAN ID that is associated with the particular VLAN. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 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.
Administrator’s Guide for SIP-T2xP IP Phones Example network.vlan.dhcp_enable = 1 Parameter- Configuration File network.vlan.dhcp_option .cfg Description Configures the DHCP option used to request the VLAN ID. Format String Default Value 132 Range 128 to 254 Example network.vlan.dhcp_option = 132 Parameter- Configuration File network.vpn_enable .cfg VPN Enables or disables VPN feature on the IP phone.
Appendix QoS 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.rtptos = 46 Parameter- Configuration File network.qos.signaltos .cfg Configures the DSCP for SIP packets.
Administrator’s Guide for SIP-T2xP IP Phones Format Boolean Default Value 0 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. Format IP Address or Domain Name Default Value Blank Range Not Applicable Example account.1.nat.stun_server = 192.168.1.20 Parameter- Configuration File account.x.
Appendix Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example network.snmp.enable = 1 Parameter- Configuration File network.snmp.port .cfg Configures the port used for SNMP communication. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value Blank Range 1 to 65535 Example network.snmp.port = 161 Parameter- Configuration File network.snmp.trust_ip .
Administrator’s Guide for SIP-T2xP IP Phones 802.1X 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 phone will reboot to make the change take effect. It is only applicable to EAP-MD5, PEAP-MSCHAPv2 and EAP-TTLS/EAP-MSCHAPv2 protocols. Format String Default Value Blank Range Not Applicable Example network.802_1x.md5_password = admin123 Parameter- Configuration File network.802_1x.root_cert_url .cfg Configures the access URL of the root certificate used for authentication.
Administrator’s Guide for SIP-T2xP IP Phones Default Value Blank Range Not Applicable Example network.802_1x.client_cert_url = http://192.168.1.10/ client.pem TR-069 Parameter- Configuration File managementserver.enable .cfg Enables or disables TR-069 feature on the IP phone. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 0 Range 0-Disabled 1-Enabled Example managementserver.
Appendix with the ACS. This string is set to the empty string if no authentication is required. Note: If you change this parameter, the phone will reboot to make the change take effect. Format String Default Value Blank Range Not Applicable Example managementserver.password = pwd123 Parameter- Configuration File managementserver.url .cfg Configures the URL of the ACS. Description Note: If you change this parameter, the phone will reboot to make the change take effect.
Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File managementserver.connection_r .cfg equest_password Configures the password for the IP phone to authenticate the incoming connection Description requests. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format String Default Value Blank Range Not Applicable Example managementserver.
Appendix phone will reboot to make the change take effect. Format Integer Default Value 60 Range Not Applicable Example managementserver.periodic_inform_interv al = 60 IPv6 Parameter- Configuration File network.ip_address_mode .cfg Configures the IP address mode. 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-IPv4 1-IPv6 2-IPv4&IPv6 Example network.
Administrator’s Guide for SIP-T2xP IP Phones Example network.ipv6_internet_port.type = 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 configured as Static IP Address and the IP address mode is configured as IPv6 or IPv4&IPv6. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank Range Not Applicable Example network.ipv6_internet_port.gateway = 3036:1:1:c3c7:c11c:5447:23a6:255 Parameter- Configuration File network.ipv6_primary_dns .
Administrator’s Guide for SIP-T2xP IP Phones Format IP Address Default Value Blank Range Not Applicable network.ipv6_secondary_dns = Example 2026:1234:1:1:c3c7:c11c:5447:23a6 Parameter- Configuration File network.ipv6_icmp_v6.enable .cfg Enables or disables ICMPv6 feature. If it is set to 1 (enabled), the IP phone obtains network settings of the IPv6 from Description the ICMPv6 protocol. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Appendix Range Example 0-Disabled 1-Enabled features.headset_prior = 1 Dual Headset Parameter- Configuration File features.headset_training .cfg Enables or disables dual headset feature. If set to 1 (Enabled), users can use two headsets on one phone. When the IP Description 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.
Administrator’s Guide for SIP-T2xP IP Phones When Y=7, the default value is 0; When Y=8, the default value is 0; When Y=9, the default value is 0; When Y=10, the default value is 0; When Y=11, the default value is 0. Range 0-Disabled 1-Enabled Example account.1.codec.1.enable = 1 Parameter- Configuration File account.x.codec.y.payload_type .cfg Configures the codec for account x to use. Description X ranges from 1 to 6. Y ranges from 1 to 11.
Appendix 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. Y ranges from 1 to 11.
Administrator’s Guide for SIP-T2xP IP Phones When Y=6, the default value is 9; When Y=7, the default value is 102; When Y=8, the default value is 112; When Y=9, the default value is 102; When Y=10, the default value is 99; When Y=11, the default value is 104. 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.
Appendix Voice Activity Detection Parameter- Configuration File voice.vad .cfg Description Enables or disables VAD feature on the IP phone. Format Boolean Default Value 0 Range Example 0-Disabled 1-Enabled voice.vad = 1 Comfort Noise Generation Parameter- Configuration File voice.cng .cfg Description Enables or disables CNG feature on the IP phone. Format Boolean Default Value 1 Range Example 0-Disabled 1-Enabled voice.
Administrator’s Guide for SIP-T2xP IP Phones 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 Not Applicable Example voice.jib.min = 60 Parameter- Configuration File voice.jib.max .cfg Configures the maximum delay time for Description jitter buffer.
Appendix Security Feature Parameters TLS Parameter- Configuration File account.x.transport .cfg Configures the transport type for account x. Description If set to 2 (TLS), the SIP message of this 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.transport = 2 Parameter- Configuration File security.trust_certificates .
Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File security.ca_cert .cfg Configures the type of certificates the IP phone used to authenticate the Description connecting server. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Boolean Default Value 0 0-Default certificates Range 1-Custom certificates 2-All certificates Example security.ca_cert = 0 Parameter- Configuration File security.
Appendix Format Boolean Default Value 0 Range Example 0-Default certificates 1-Custom certificates security.dev_cert = 0 Uploading Certificates Parameter- Configuration File trusted_certificates.url .cfg Configures the access URL of the certificate used to authenticate the connecting Description server. Note: The certificate you want to upload must be in *.pem, *.crt, *.cer or *.der format.
Administrator’s Guide for SIP-T2xP IP Phones SRTP Parameter- Configuration File account.x.srtp_encryption .cfg Configures whether to use voice encryption service. If the 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 set to 2 (Compulsory), the IP phone is forced to using SRTP during a call. X ranges from 1 to 6.
Appendix Value 0-Disabled 1-Enabled Example auto_provision.aes_key_in_file = 0 Parameter- Configuration File auto_provision.aes_key_16.com .cfg Configures the plaintext AES key which is used to decrypt the .cfg Description file. Note: It works only if the parameter “auto_provision.aes_key_in_file” is set to 0 (Disabled). Format String Default Value Blank Range Example 16 characters and the supported characters contain: 0 ~ 9, A ~ Z, a ~ z. auto_provision.
Administrator’s Guide for SIP-T2xP IP Phones Upgrading Firmware Parameter- Configuration File auto_provision.mode .cfg Description Configures the auto provision mode. Format Integer Default Value 1 Valid values are: 0-Disabled 1-Power on (when the IP phone reboots) Range 4-Repeatedly (at a fixed interval) 5-Weekly (at the specified time) 6-Power on + Repeatedly 7-Power on + Weekly Example auto_provision.mode = 1 Parameter- Configuration File auto_provision.schedule.
Appendix Note: It works only if the parameter “auto_provision.mode” is set to 5(Weekly) or 7 (Power on + Weekly). Format 00:00 Default Value 00:00 Range 00:00 to 23:59 Example auto_provision.schedule.time_from = 01:30 Parameter- Configuration File auto_provision.schedule.time_to < y0000000000xx >.cfg Configures the end time of day in 24-hour period for the IP phone to check new Description configuration files. Note: It works only if the parameter “auto_provision.
Administrator’s Guide for SIP-T2xP IP Phones 3-Wednesday 4-Thursday 5-Friday 6-Saturday Example auto_provision.schedule.dayofweek = 0123456 Parameter- Configuration File firmware.url .cfg Description Configures the access URL of the firmware. Format String Default Value Blank Range Not Applicable Example firmware.url = http://192.168.1.20/2.71.0.140.rom Resource Files Access URL of Replace Rule Template Parameter- Configuration File dialplan_replace_rule.
Appendix Access URL of Dial-now Template Parameter- Configuration File dialplan_dialnow.url .cfg Description Configures the access URL of the dial-now template. Format URL Default Value Blank Range Not Applicable Example dialplan_dialnow.url = http://192.168.10.25/dialnow.xml Access URL of Softkey Layout Template Parameter- Configuration File custom_softkey_call_failed.url .
Administrator’s Guide for SIP-T2xP IP Phones Format URL Default Value Not Applicable Range Not Applicable 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 The following example uses HTTP to download the Dialing state file from the Example “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. 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 Description customized file for the soft key presented on the LCD screen when in the RingBack state.
Administrator’s Guide for SIP-T2xP IP Phones custom_softkey_talking.url = http://10.2.8.16:8080/XMLfiles/Talking.xml Access URL of Local Contact File Parameter- Configuration File local_contact.data.url .cfg Description Configures the access URL of the local contact file. Format URL Default Value Blank Range Not Applicable Example local_contact.data.url = http://192.168.10.25/contactData1.xml Access URL of Remote XML Phone Book Parameter- Configuration File remote_phonebook.
Appendix server where to export the log files. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank Range Not Applicable Example syslog.server = 192.168.1.50 Parameter- Configuration File syslog.log_level .cfg Configures the severity level of the logs to be reported to a log file. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Administrator’s Guide for SIP-T2xP IP Phones Configuring DSS Key This section provides the DSS key parameters you can configure on the IP phone. DSS key consists of memory key and line key. The following table lists the number of DSS keys you can configure for each phone model: Phone Model Line Key Memory Key T28P 6 10 T26P 3 10 T22P 3 / T20P 2 / DSS key can be assigned with various key features.
Appendix Format Default Value URL (not applicable to SIP-T20P) Group Listening XML Group (not applicable to SIP-T20P) Group Pickup Multicast Paging Record XML Browser (not applicable to SIP-T20P) URL Record LDAP (not applicable to SIP-T20P) Prefix Zero Touch ACD Hot Desking Local Group Keypad Lock Custom Button (not applicable to SIP-T20P) Directory Integer For the memory key, the default value is 0 (N/A).
Administrator’s Guide for SIP-T2xP IP Phones 27-XML Browser 34-Hot Desking 35-URL Record 38-LDAP 40-Prefix 41-Zero Touch 42-ACD 45-Local Group 48-Custom Button 50-Keypad Lock 61-Directory Example memorykey.1.type = 8 Parameter- Configuration File memorykey.x.line .cfg ParameterLine key. 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 ACD Hot Desking Zero Touch URL (not applicable to the SIP-T20P IP phone) Format Keypad Lock Directory Integer For the memory 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. Valid values are: 0 to 6 (for T28P) 0 to 3 (for T26P/T22P) 0 to 2 (for T20P) Range 0-Line 1 1-Line 1 2-Line 2 … 6-Line 6 Example memorykey.1.
Administrator’s Guide for SIP-T2xP IP Phones the number you want to dial out. memorykey.1.value = 1001 Parameter- Configuration File memorykey.x.pickup_value .cfg Parameterlinekey.x.pickup_value Configures the pickup code for BLF feature. Description This parameter is only applicable to BLF feature. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example memorykey.1.
Appendix … Format Integer Default Value 0 Range Not Applicable Example Specify the second remote phone book. memorykey.1.xml_phonebook = 1 Keypad Lock Key Parameter- Configuration File memorykey.x.type .cfg Parameterlinekey.x.type Configures a DSS key to be Keypad Lock key on the IP phone. Description The digit 50 stands for the key type Keypad Lock. 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 IP Phones Value 5 Example memorykey.1.type = 5 Directed Call Pickup Key Parameter- Configuration File memorykey.x.type .cfg Parameterlinekey.x.type Configures a DSS key to be directed call pickup key on the IP phone. Description The digit 9 stands for the key type Call Pickup. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Value 9 Example memorykey.1.
Appendix 6-Line 6 Example memorykey.1.line = 1 Parameter- Configuration File memorykey.x.value .cfg Parameterlinekey.x.value Configures the directed call pickup feature code followed by the number of monitored Description extension. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format String Range Not Applicable Example memorykey.1.value = *971001 Group Call Pickup Key Parameter- Configuration File memorykey.x.type .
Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File memorykey.x.line .cfg Parameterlinekey.x.line Configures the desired line to apply the group Description call pickup key. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Valid values are: 0 to 6 (for T28P) 0 to 3 (for T26P/T22P) 0 to 2 (for T20P) Range 0-Line 1 1-Line 1 2-Line 2 … 6-Line 6 Example memorykey.1.
Appendix Call Return Key Parameter- Configuration File memorykey.x.type .cfg Parameterlinekey.x.type Configures a DSS key to be call return key on the IP phone. Description The digit 7 stands for the key type Call Return. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Value 7 Example memorykey.2.type = 7 Call Park Key Parameter- Configuration File memorykey.x.type .cfg Parameterlinekey.x.
Administrator’s Guide for SIP-T2xP IP Phones Parameter- Configuration File memorykey.x.line .cfg Parameterlinekey.x.line Configures the desired line to apply key feature. Description For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Valid values are: 0 to 6 (for T28P) 0 to 3 (for T26P/T22P) 0 to 2 (for T20P) Range 0-Line 1 1-Line 1 2-Line 2 … 6-Line 6 Example memorykey.2.line = 0 Parameter- Configuration File memorykey.x.
Appendix Intercom Key Parameter- Configuration File memorykey.x.type .cfg Parameterlinekey.x.type Configures a DSS key to be the intercom key. Description The digit 14 stands for the key type Intercom. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Value 14 Example memorykey.2.type = 14 Parameter- Configuration File memorykey.x.line .cfg Parameterlinekey.x.
Administrator’s Guide for SIP-T2xP IP Phones 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. Format String Range Not Applicable Example memorykey.2.value = 1008 LDAP Key Parameter- Configuration File memorykey.x.type .cfg Parameterlinekey.x.type Configures a DSS key to be LDAP key on the IP phone.
Appendix For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Value 16 Example memorykey.3.type = 16 Parameter- Configuration File memorykey.x.line .cfg Parameterlinekey.x.line Configures the desired line to apply the BLF key. Description 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 IP Phones Example memorykey.3.value = 1008 Parameter- Configuration File memorykey.x.pickup_value .cfg Parameterlinekey.x.pickup_value Configures the pickup code for the BLF feature. Description This parameter only applies to the BLF feature. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example memorykey.3.
Appendix Multicast Paging Key Parameter- Configuration File memorykey.x.type .cfg Parameterlinekey.x.type Configures a DSS key to be a multicast paging key on the IP phone. Description The digit 24 stands for the key type Multicast Paging. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Value 24 Example memorykey.2.type = 24 Parameter- Configuration File memorykey.x.value .cfg Parameterlinekey.x.
Administrator’s Guide for SIP-T2xP IP Phones Record Key Parameter- Configuration File memorykey.x.type .cfg Parameterlinekey.x.type Configures a DSS key to be a record key on the IP phone. Description The digit 25 stands for the key type Record. For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format Integer Value 25 Example memorykey.2.type = 25 URL Record Key Parameter- Configuration File memorykey.x.type .
Appendix For the memory key, x ranges from 1 to 10. For the line key, x ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example memorykey.1.value = http://10.1.2.224/phonerecording.cgi Hot Desking Key Parameter- Configuration File memorykey.x.type .cfg Parameterlinekey.x.type Configures a DSS key to be a hot desking key on the IP phone. Description The digit 34 stands for the key type Hot Desking. For the memory key, x ranges from 1 to 10.
Administrator’s Guide for SIP-T2xP IP Phones RFC and Internet Draft Support The following RFC’s and Internet drafts are supported: RFC 1321—The MD5 Message-Digest Algorithm RFC 2327—SDP: Session Description Protocol RFC 2387—The MIME Multipart / Related Content-type RFC 2976—The SIP INFO Method RFC 3261—SIP: Session Initiation Protocol (replacement for RFC 2543) RFC 3262—Reliability of Provisional Responses in the Session Initiation Protocol (SIP) RFC 3263—Session Initiation Prot
Appendix draft-anil-sipping-bla-02.txt—Implementing Bridged Line Appearances (BLA) Using Session Initiation Protocol (SIP) draft-ietf-sip-privacy-04.txt—SIP Extensions for Network-Asserted Caller Identity and Privacy within Trusted Networks draft-levy-sip-diversion-06.txt—Diversion Indication in SIP draft-ietf-sipping-cc-conferencing-03.txt—SIP Call Control - Conferencing for User Agents draft-ietf-sipping-rtcp-summary-02.
Administrator’s Guide for SIP-T2xP IP Phones Method Supported UPDATE Yes PUBLISH Yes Notes SIP Header The following SIP request headers are supported: Method 396 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-Asserted-Identity Yes P-Preferred-Identity Yes Proxy-Authentic
Appendix Method Supported 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 SIP Responses The following SIP responses are supported: 1xx Response—Information Responses 1xx Response Supported 100 Trying Yes 180 Ringing Yes 181 Call Is Being Forwarded Yes 183 Session Progress Yes Notes 2xx Response—Successful Responses 2xx Response S
Administrator’s Guide for SIP-T2xP IP Phones 3xx Response—Redirection Responses 3xx Response Supported 300 Multiple Choices Yes 301 Moved Permanently Yes 302 Moved Temporarily Yes Notes 4xx Response—Request Failure Responses 4xx Response 400 Bad Request Yes 401 Unauthorized Yes 402 Payment Required Yes 403 Forbidden Yes 404 Not Found Yes 405 Method Not Allowed Yes 406 Not Acceptable No 407 Proxy Authentication Required Yes 408 Request Timeout Yes 409 Conflict No 410 Gone No 4
Appendix 4xx Response Supported 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 Notes 5xx Response—Server Failure Responses 5xx Response Supported 500 Internal Server Error Yes 501 Not Implemented Yes 502 Bad Gateway No 503 Service Unavailable No 504 Gateway Timeout No 505 Version Not Supported No Notes 6xx Response—
Administrator’s Guide for SIP-T2xP IP Phones o—Owner/creator and session identifier Yes a—Media attribute Yes c—Connection information Yes m—Media name and transport address Yes s—Session name Yes t—Active time Yes Appendix E: SIP Call Flows 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 Successful Call Setup and Disconnect The following figure illustrates the scenario of a successful call. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones. The call flow scenario is as follows: 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.
Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends a SIP INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted 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 B sends a SIP 200 OK response to F7 200 OK— User B to Proxy the proxy server. The 200 OK response Server notifies User A that the connection has been made. The proxy server forwards the 200 OK F8 200OK—Proxy Server to User A message to User A. The 200 OK 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.
Administrator’s Guide for SIP-T2xP 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.
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 IP Phones Step F6 Action 486 Busy Here—Proxy Server to User A Description The proxy server forwards the 486 Busy Here response to notify User A that User B is busy. User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server. The SIP ACK message indicates that User A has received the 486 Busy Here message.
Appendix Unsuccessful Call Setup—Called User Does Not Answer The following figure illustrates the scenario of an unsuccessful call due to the reason of the called user not answering the call. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones. The call flow scenario is as follows: 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.
Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted 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 B User A wants to disconnect the call. 200 OK—User B to Proxy Server User B sends a SIP 200 OK response to the proxy server. The SIP 200 OK F7 response indicates that User B has received the CANCEL request. F8 200 OK—Proxy Server to User The proxy server forwards the SIP 200 OK A response to notify User A that the CANCEL request has been processed successfully.
Administrator’s Guide for SIP-T2xP IP Phones Successful Call Setup and Call Hold The following figure illustrates a successful call setup and call hold. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones. 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. 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.
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 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 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 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.
Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted 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 C sends a SIP INVITE message to the proxy server.
Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends a mid-call INVITE request to F13 INVITE—User A to Proxy Server the proxy server with new SDP session parameters, which are used to place the call on hold. F14 INVITE—Proxy Server to User B The proxy server forwards the mid-call INVITE message to User B. User B sends a 200 OK to the proxy F15 200 OK—User B to Proxy Server server. The 200 OK response indicates that the INVITE was successfully processed.
Appendix Call Transfer without Consultation 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 consulting the third party. 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: 1. User A calls User B.
Administrator’s Guide for SIP-T2xP IP Phones User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9. 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.
Appendix Step Action Description User A sends an INVITE message to the proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted 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 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 B sends a REFER message to the F9 REFER—User B to Proxy Server proxy server.
Appendix Step Action F18 INVITE—Proxy Server to User C Description The proxy server maps the SIP URI in the To field to User C. User C sends a SIP 180 Ringing response F19 180 Ringing—User C to Proxy to the proxy server. The 180 Ringing Server response indicates that the user is being alerted. The proxy server forwards the 180 F20 180 Ringing—Proxy Server to Ringing response to User A.
Administrator’s Guide for SIP-T2xP IP Phones Call is established between User B and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. 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.
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 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 Description sends the INVITE request to User C. User C sends a SIP 180 Ringing response F17 180 Ringing—User C to Proxy to the proxy server. The 180 Ringing Server response indicates that the user is being alerted. The proxy server forwards the 180 F18 180 Ringing—Proxy Server to Ringing response to User A. User A hears User A the ring-back tone indicating that User C is being alerted.
Administrator’s Guide for SIP-T2xP IP Phones Step Action Description response indicates that User B accepts the transfer. User A terminates the call session by 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 Server the proxy server.
Appendix Always Call Forward The following figure illustrates successful call forwarding between Yealink SIP IP phones in which User B has enabled always call forward. The incoming call is immediately forwarded to User C when User A calls User B. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: 1.
Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of the User B is 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 INVITE request to the proxy server. In the INVITE request, a F7 INVITE—User A to Proxy Server unique Call-ID is generated and the Contact-URI field indicates that User A requested the call. The proxy server maps the SIP URI in the F8 INVITE—Proxy Server to User C To field to User C. The proxy server sends the SIP INVITE request to User C. User C sends a SIP 180 Ringing response F9 180 Ringing—User C to Proxy to the proxy server.
Administrator’s Guide for SIP-T2xP IP Phones Busy Call Forward 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.
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 IP Phones Step Action Description 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 proxy server.
Appendix No Answer Call Forward 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.
Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted 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 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 proxy server.
Administrator’s Guide for SIP-T2xP IP Phones Call Conference The following figure illustrates successful 3-way calling between Yealink SIP-T2xP 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.
Appendix User A User B Proxy Server F1. INVITE B F4. 180 Ringing F6. 200 OK F7. ACK User C 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. ACK Session 1 established between User A and User B is hold F15. INVITE C F16. INVITE C F17. 180 Ringing F18. 180 Ringing F20. 200 OK F19. 200 OK F21. ACK F22.
Administrator’s Guide for SIP-T2xP IP Phones Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted 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 IP Phones Step Action Description the SIP INVITE request to User C. User C sends a SIP 180 Ringing response F17 180 Ringing—User C to Proxy to the proxy server. The 180 Ringing Server response indicates that the user is being alerted. The proxy server forwards the 180 F18 180 Ringing—Proxy Server to Ringing response to User A. User A hears User A the ring-back tone indicating that User C is being alerted.
Appendix Appendix F: Sample Configuration File This section provides the sample configuration file necessary to configure the IP phone. Any line starts with a pound sign (#) is considered to be a comment, unless the # is contained within double quotes. For Boolean fields, 0 = disabled, 1 = enabled. This file contains sample configurations for the .cfg or .cfg file. The parameters included here are examples only. Not all possible parameters are shown in the sample configuration file.
Administrator’s Guide for SIP-T2xP IP Phones #Time Settings local_time.time_zone = local_time.time_zone_name = local_time.ntp_server1 = local_time.ntp_server2 = local_time.interval = local_time.dhcp_time = #Use the following parameters to set the time and date manually. local_time.manual_time_enable = local_time.date_format = local_time.time_format = #Auto DST Settings local_time.summer_time = local_time.dst_time_type = local_time.start_time = local_time.end_time = local_time.
Appendix #Hotline features.hotline_number = features.hotline_delay = #Web Server Type network.web_server_type = network.port.http = network.port.https = #DTMF Suppression features.dtmf.hide = features.dtmf.hide_delay = #Call Forward # In Phone Mode features.fwd_mode = 0 forward.always.enable = forward.always.target = forward.always.on_code = forward.always.off_code = forward.busy.enable = forward.busy.target = forward.busy.on_code = forward.busy.off_code = forward.no_answer.enable = forward.no_answer.
Administrator’s Guide for SIP-T2xP IP Phones account.1.timeout_fwd.off_code = #Call Transfer transfer.semi_attend_tran_enable = transfer.blind_tran_on_hook_enable = transfer.on_hook_trans_enable = transfer.tran_others_after_conf_enable = #Call Conference account.1.conf_type = account.1.conf_uri = #DTMF account.1.dtmf.type = account.1.dtmf.dtmf_payload = account.1.dtmf.info_type = #Distinctive Ring Tones account.1.alert_info_url_enable = distinctive_ring_tones.alert_info.1.text = distinctive_ring_tones.
Appendix ldap.base = ldap.user = ldap.password = ldap.max_hits = ldap.name_attr = ldap.numb_attr = ldap.display_name = ldap.version = ldap.call_in_lookup = ldap.ldap_sort = #Action URL action_url.setup_completed = action_url.log_on = action_url.log_off = action_url.register_failed = action_url.off_hook = action_url.on_hook = action_url.incoming_call = action_url.outgoing_call = action_url.call_established = action_url.dnd_on = action_url.dnd_off = action_url.always_fwd_on = action_url.
Administrator’s Guide for SIP-T2xP IP Phones action_url.transfer_failed = #SNMP network.snmp.enable = network.snmp.port = network.snmp.trust_ip = #Access URL of Resource Files dialplan_dialnow.url = dialplan_replace_rule.url = local_contact.data.url = remote_phonebook.data.1.
Index Index Numeric C 180 Ring Workaround 87 Call Completion 802.
Administrator’s Guide for SIP-T2xP IP Phones Early Media 87 Missed Call Log Encrypting Configuration Files 215 Enabling the Watch Dog Feature 235 65 Multicast Paging 147 Music on Hold 142 G N Getting Information from Status Indicators 236 NAT Traversal Getting Started Network Address Translation (NAT) 11 Group Call Pickup 106 180 Network Conference 180 101 No Answer Forward 95 H H.
Index Softkey Layout 57 Specifying the Language to Use SRTP 53 213 STUN Server 180 Suppress DTMF Display 121 Summary of Changes vi T Table of Contents Time and Date xi 47 Transfer on Conference Hang Up Transfer via DTMF 102 122 Transport Layer Security (TLS) Troubleshooting 207 231 Troubleshooting Methods 231 Troubleshooting Solutions 237 TR-069 Device Management 189 U Upgrading Firmware 219 Use Outbound Proxy in Dialog User Agent Client (UAC) 2 User Agent Server (UAS) 3 User
