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.
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.
About This Guide The guide is considered to be an administration-level version, which is intended for administrators who need to properly configure, customize, manage, and troubleshoot the IP phone systems rather than the end-users of IP phones. It provides details on the functionality and configuration of the IP phones. Many of the features are described in this guide involving the network settings, which could affect the IP phones’ performance in the network.
Administrator’s Guide for SIP-T46G IP Phone Chapter 3, ―Configuring Basic Features‖ describes how to configure the basic 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.
Table of Contents About This Guide ...................................................................... v Documentations ............................................................................................................................... v In This Guide .................................................................................................................................... v Table of Contents ....................................................................
Administrator’s Guide for SIP-T46G IP Phone User Password ............................................................................................................................... 36 Administrator Password ................................................................................................................ 37 Phone Lock ..................................................................................................................................... 38 Date and Time .....................
Table of Contents Intercom........................................................................................................................................ 110 Outgoing Intercom Calls...................................................................................................... 110 Incoming Intercom Calls ...................................................................................................... 111 Configuring Advanced Features...........................................
Administrator’s Guide for SIP-T46G IP Phone Transport Layer Security.............................................................................................................. 193 Secure Real-Time Transport Protocol .......................................................................................... 199 Encrypting Configuration Files ................................................................................................... 201 Upgrading the Firmware.......................................
Table of Contents Appendix B: Time Zones ............................................................................................................. 229 Appendix C: Configuration Parameters .................................................................................... 232 Setting Parameters in Configuration Files .......................................................................... 232 Basic and Advanced Parameters ...............................................................................
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Product Overview This chapter contains the following information about the SIP-T46G IP phones: VoIP Principle SIP Components Introducing the SIP-T46G IP Phones 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-T46G IP Phone SIP provides the capabilities to: Determine the location of the target endpoint -- SIP supports address resolution, name mapping, and call redirection. Determine the media capabilities of the target endpoint -- Via Session Description Protocol (SDP), SIP determines the ―lowest level‖ of common services between the endpoints. Conferences are established using only the media capabilities that can be supported by all endpoints.
Product Overview method may be the preferred measure when not using an application layer firewall, application layer firewalls like to know what applications are flowing though which ports and it is possible using content types of other applications other than the one you are trying to let through which has been denied.
Administrator’s Guide for SIP-T46G IP Phone SIP-T46G Physical Features: 4 - 4.3‖ TFT-LCD, 480 x 272 pixel, 16.
Product Overview In addition to the physical features introduced above, the SIP-T46G 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 conference. - Basic Features: DND, phone lock, auto redial, live dialpad, dial plan, hotline, caller identity, auto answer.
Administrator’s Guide for SIP-T46G IP Phone 6 - SRTP (RFC3711) - Transport Layer Security (TLS) - VLAN (802.
Getting Started This chapter introduces the initialization of the SIP-T46G IP phones, the installing and connecting process of the IP phones which you need to follow.
Administrator’s Guide for SIP-T46G IP Phone 1) Attach the stand: Desk Mount Method Wall Mount Method 8 2) Connect the handset and optional headset: 3) Connect the network and power: AC power Power over Ethernet (PoE)
Getting Started 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 phones and connect the other end of the power adapter into an electrical power outlet. 2. Connect the supplied Ethernet cable between the Internet port on the IP phones and the Internet port in your network or switch/hub device port. Power over Ethernet Using a regular Ethernet cable, the IP phones can be powered from a PoE (IEEE 802.3af) compliant switch or hub.
Administrator’s Guide for SIP-T46G IP Phone The initialization process of the IP phones is responsible for network connectivity and operation of the IP phones in your local network. Once you connect your IP phone to the network and to an electrical supply, the IP phone begins its initialization process. During the initialization process, the following events proceed: Loading the ROM file The ROM file resides in the flash memory of the IP phones.
Getting Started Updating the firmware If the access URL of the firmware is defined in the configuration file, the IP phone will download the firmware from the provisioning server. If the MD5 value of the downloaded firmware file differs from that of the image stored in the flash memory, the IP phone performs a firmware update. Downloading the resource files In addition to configuration file(s), the IP phones may require resource files before it can deliver service.
Administrator’s Guide for SIP-T46G IP Phone An administrator or a user can configure and use the IP phones via phone user interface. Specific features access is restricted to the administrator. These specific features are password protected by default. The default password is ―admin―(case-sensitive). Not all features are available on configuring via phone user interface. An administrator or a user can configure the IP phones via web user interface.
Getting Started Each line in a configuration file must use the following format and adhere to the following rules: variable-name = value - Associate only one value with one variable. - Separate variable name and value with equal sign. - Set only one variable per line. - Put the variable and value on the same line, and do not break the line. - Comment the variable on a separated line. Use the pound (#) delimiter to distinguish the comments.
Administrator’s Guide for SIP-T46G IP Phone Icons Description Numeric input mode Voice Mail Text Message Auto Answer Do Not Disturb Call Forward Call Hold Call Mute Ringer volume is 0 Keypad Lock Received Calls Dialed Calls Missed Calls Forwarded Calls Recording box is full A call cannot be recorded Recording starts successfully Recording cannot be started Recording cannot be stopped Open VPN Bluetooth 14
Getting Started Icons Description Bluetooth headset is both paired and connected Conference The contact icon The default contact photo This section describes how to configure the basic network parameters that are required for the IP phones to operate in the network. DHCP (Dynamic Host Configuration Protocol) is a network protocol used to dynamically allocate network parameters to hosts connected to a network.
Administrator’s Guide for SIP-T46G IP Phone Parameter DHCP Option Router 3 Time Server 4 Domain Name Server 6 Log Server 7 Host Name 12 Domain Server 15 Broadcast Address 28 Network Time Protocol 42 Servers Vendor-Specific Information Vendor Class Identifier TFTP Server Name Description Specify a list of IP addresses for routers on the client’s subnet. Specify a list of time servers available to the client. Specify a list of domain name servers available to the client.
Getting Started Phone User Interface Configure DHCP on the IP phone. To configure DHCP via web user interface: 1. Click on Network->Basic. 2. In the IPv4 Config block, mark the DHCP radio box. 3. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 4. Click OK to reboot the IP phone. To configure DHCP via phone user interface: 1. Press Menu->Advanced (password: admin) ->Network->WAN Port->IPv4. 2. Press 3.
Administrator’s Guide for SIP-T46G IP Phone Secondary DNS Procedure Network parameters can be configured manually using the configuration files or locally. Configure network parameters of the IP phone manually. Configuration File .cfg For more information, refer to Static Network Settings on page 233. Configure network parameters of the IP phone manually.
Getting Started 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 IP address, subnet mask, default gateway, primary DNS and secondary DNS in the corresponding fields. 4. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 5. Click OK to reboot the IP phone. To configure the IP address mode via phone user interface: 1.
Administrator’s Guide for SIP-T46G IP Phone The IP phone reboots automatically to make the settings effective after a period of time. Note Using the wrong network parameters may result in inaccessibility of your phone and may also have an impact on your network performance. For more information on these parameters, contact your network administrator.
Getting Started 3. Enter the username and password in the corresponding fields. 4. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 5. Click OK to reboot the IP phone. To configure PPPoE via phone user interface: 1. Press Menu->Advanced (password: admin) ->Network->WAN Port->IPv4. 2. Press 3. Enter the username and password in the corresponding fields. 4. Press the Save soft key to accept the change.
Administrator’s Guide for SIP-T46G IP Phone transmission parameters (e.g., speed and duplex mode) to transmit voice or data over Ethernet. In this process, the connected devices first share transmission capabilities and then choose the highest performance transmission mode they both support. You can configure the Internet port and PC port on the IP phones to auto-negotiate during the transmission.
Getting Started Procedure The transmission method of Ethernet port can be configured using the configuration files or locally. Configure the transmission method of Ethernet port. Configuration File .cfg For more information, refer to Internet and PC Ports Negotiation on page 236. Configure the transmission method of Ethernet port.
Administrator’s Guide for SIP-T46G IP Phone to search and manipulate text based on patterns. Regular expression can be used to define dial plan for the IP phones. Dial plan is a string of characters that governs the way for the IP phones processing the inputs received from the IP phone keypads. The IP phones support the following dial plan features: Replace Rule Dial-now Area Code Block Out The priority of matching dial plan is: Dial-now>Replace Rule>Area Code>Block Out.
Getting Started Replace rule is an alternative string that replaces the numbers entered by the user. You can create up to 20 replace rules for the IP phones. The replace rules can be created either one by one or in batch using a replace rule template. For more information on the replace rule template, refer to Replace Rule Template on page 209. Procedure Replace rule can be created using the configuration files or locally. Create the replace rule for the IP Configuration File .cfg phone.
Administrator’s Guide for SIP-T46G IP Phone 5. Click Add to add the replace rule. Dial-now is a string used to match the numbers entered by the user. When entered numbers match the predefined dial-now rule, the IP phones will automatically dial out the numbers without pressing the send key. You can create up to 20 dial-now rules for the IP phones. The dial-now rules can be created either one by one or in batch using a dial-now rule template.
Getting Started If you leave the field blank or enter an invalid value, the dial-now rule applies to all accounts on the IP phone. 4. Click Add to add the dial-now rule. To configure the delay time for the dial-now rule via web user interface: 1. Click on Features->General Information. 2. Enter the desired time within 1-14 (in seconds) in the Time Out for Dial-Now Rule field. 3. Click Confirm to accept the change. Area codes are also known as Numbering Plan Areas (NPAs).
Administrator’s Guide for SIP-T46G IP Phone code rule, the IP phones will automatically add the area code to the beginning of the numbers and dial out. The IP phones only support one area code rule. Procedure Area code rule can be configured using the configuration files or locally. Create the area code rule and specify the maximum and Configuration File .cfg minimum lengths of the entered numbers. For more information, refer to Dial Plan on page 237.
Getting Started Block out rule can prevent users from dialing out some specific numbers. When entered numbers match the predefined block out rule, the phone LCD screen prompts ―Forbidden Number‖. You can create up to 10 block out rules. Procedure Block out rule can be created using the configuration files or locally. Create the block out rule for the Configuration File .cfg IP phone. For more information, refer to Dial Plan on page 237. Create the block out rule for the desired line.
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Configuring Basic Features This chapter provides information for making configuration changes for the following basic features: Wallpaper Backlight User Password Administrator Password Phone Lock Date and Time Language Softkey Layout Key as Send Hotline Call Log Missed Call Log Local Directory Live Dialpad Call Waiting Auto Redial Auto Answer Call Completion Anonymous Call Anonymous Call Rejection Do Not Disturb Busy Ton
Administrator’s Guide for SIP-T46G IP Phone 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 Wallpaper is an image used as the background of the phone idle screen.
Configuring Basic Features wallpaper. Change the wallpaper via web user interface. Navigate to: http:///servlet ?p=settings-preference&q=load Phone User Interface Change the wallpaper via phone user interface. To upload a customized wallpaper via web user interface: 1. Click on Settings->Preference. 2. In the Upload Wallpaper field, click Browse to select the wallpaper image from your local system. 3. Click Upload to upload the file. 4. Click Confirm to accept the change.
Administrator’s Guide for SIP-T46G IP Phone 2. Select the desired wallpaper from the pull-down list of Wallpaper. 3. Click Confirm to accept the change. To change the wallpaper via phone user interface: 1. Press Menu->Basic->Display->Wallpaper. 2. Press 3. Press the Save soft key to accept the change. or , or the Switch soft key to select the desired wallpaper. Backlight provides the brightness necessary for making the phone LCD screen readable in a darkened environment.
Configuring Basic Features Backlight on page 241. Configure the backlight of the LCD screen. Web User Interface Navigate to: http:///servlet Local ?p=settings-preference&q=load Phone User Interface Configure the backlight of the LCD screen. To configure the backlight via web user interface: 1. Click on Settings->Preference. 2. Select the desired value from the pull-down list of Backlight Idle Intensity. 3. Select the desired value from the pull-down list of Backlight On Intensity.
Administrator’s Guide for SIP-T46G IP Phone Several menu options are protected with two privilege levels, user and administrator, each with its own password. When logging in the web user interface, you need to enter the username and password for granting access to various menu options. A user or an administrator can change the user password. IP phones support ASCII characters 32-126(0x20-0x7E) only in passwords.
Configuring Basic Features Advanced menu options are restricted to an administrator. Users can configure them only if they have administrator privileges. The administrator password can be only changed by the administrator. The IP phones support ASCII characters 32-126(0x20-0x7E) only in passwords. A valid password should be complex and contains at least 6 characters, where at least one character is numeric, and one character is alphabetic.
Administrator’s Guide for SIP-T46G IP Phone To change the administrator password via phone user interface: 1. Press Menu->Advanced (password: admin) ->Set Password. 2. Enter the current administrator password in the Current Password field. 3. Enter the new administrator password in the New Password field and Confirm Password field. 4. Press the Save soft key to accept the change. Phone lock is used to lock the IP phones to prevent it from unauthorized use.
Configuring Basic Features http:///servl et?p=features-phonelock&q=lo ad Assign a keypad lock key. Navigate to: http:///servl et?p=dsskey&model=1&q=loa d&linepage=1 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 Enable. 3.
Administrator’s Guide for SIP-T46G IP Phone 2. In the desired DSS key field, select Keypad Lock from the pull-down list of Type. 3. Click Confirm to accept the change. To configure the type of phone lock via phone user interface: 1. Press Menu->Advanced (password: admin) ->Phone Settings->Keypad Lock. 2. Press or , or the Switch soft key to select the desired value from the Keypad Lock Enable field. 3.
Configuring Basic Features configuring the IP phones to obtain the date and time from the NTP server, you need to set 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 near the start of spring and are adjusted backward in autumn. Many countries have used the DST at various times, details vary by location.
Administrator’s Guide for SIP-T46G IP Phone Configure the NTP server, time zone and DST. Configure the date and time manually. Web User Interface Configure the date and time formats. Navigate to: http:///servlet Local ?p=settings-datetime&q=load Configure the NTP server and time zone. Phone User Interface Configure the date and time manually. Configure the date and time formats. To configure the NTP server, time zone and DST via web user interface: 1. Click on Settings->Time & Date.
Configuring Basic Features Enter the end time in the End Date field. - Mark the DST By Week radio box in the Fixed Type field. Select the desired values from the pull-down lists of DST Start Month, DST Start Day of Week, DST Start Day of Week Last in Month, DST Stop Month, DST Stop Day of Week and DST Stop Day of Week Last in Month. 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.
Administrator’s Guide for SIP-T46G IP Phone 2. Select Enabled from the pull-down list of Manual Time. 3. Enter the date and time in the corresponding fields. 4. Click Confirm to accept the change. To configure the date and time 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.
Configuring Basic Features The default time zone is "+8 China(Beijing)". 3. Enter the domain names or IP addresses in the NTP Server 1 and NTP Server 2 fields, respectively. 4. Press or, or the Switch soft key to select Automatic from the Daylight Saving field. 5. Press the Save soft key to accept the change. To configure the date and time manually via phone user interface: 1. Press Menu->Basic->Date & Time->General->Manual. 2. Enter the specific date and time. 3.
Administrator’s Guide for SIP-T46G IP Phone All supported languages may not be available for selection. The languages available for selection depend on the language packs currently loaded on the IP phones. You can make languages available to use on the phone user interface by loading language packs to the IP phones. You can only load language packs to the IP phones using the configuration files.
Configuring Basic Features Procedure Specify the language for the web user interface or the phone user interface using the configuration files or locally. Specify the languages for the phone user interface and the Configuration File .cfg web user interface. For more information, refer to Language on page 250. Specify the language for the web user interface.
Administrator’s Guide for SIP-T46G IP Phone Softkey layout is used to customize the soft keys at the bottom of the phone LCD screen to best suit the needs of users. It can be configured based on the call states. In addition to specifying which soft keys to display, you can determine the display order of the soft keys. You can create a template about the softkey layout of the different call states. For more information on the softkey layout template, refer to Softkey Layout Template on page 211.
Configuring Basic Features Call State Default Soft Key Optional Soft Key Empty Cancel SemiAttendTransBack Transfer Empty Empty Switch Empty Cancel Talk Transfer Empty HOLD MUTE Conference SWAP Cancel NewCall Switch Answer Reject Hold Talking Held 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
Administrator’s Guide for SIP-T46G IP Phone Call State Default Soft Key Conferenced Optional Soft Key Empty Empty Hold Switch Split Answer Cancel Reject Mute Manager Procedure Softkey layout can be configured using the configuration files or locally. Specify the access URL of the softkey layout template. Configuration File .cfg For more information, refer to Access URL of Softkey Layout on page 346. Configure the softkey layout.
Configuring Basic Features To adjust the order of the soft key, click 7. or . Click Confirm to accept the change. The key as send feature allows assigning the pound key or star key as a send key. The send tone feature determines whether the IP phone plays a key tone when a user presses the send key. Procedure Key as send can be configured using the configuration files or locally. Configure the send key. Configuration File .cfg Configure the send tone feature.
Administrator’s Guide for SIP-T46G IP Phone 2. Select the desired value from the pull-down list of Key As Send. 3. Click Confirm to accept the change. To configure the send tone 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 the send key via phone user interface: 1. Press Menu->Call Feature->Others->General. 2.
Configuring Basic Features 3. Note Press the Save soft key to accept the change. The send tone feature works only if the key tone feature is enabled. The key tone feature is enabled by default. A hotline is a point-to-point communication link in which a call is automatically directed to the preset hotline number. The IP phone automatically dials out the hotline number using the first available line after a time interval when off-hook. The IP phones only support one hotline number.
Administrator’s Guide for SIP-T46G IP Phone 3. Enter the delay time in the Hotline Delay ( 0~10s) field. 4. Click Confirm to accept the change. To configure hotline via phone user interface: 1. Press Menu->Call Feature->Others->Hotline. 2. Enter the hotline number in the Number field. 3. Enter the delay time in the Hotline Delay 0-10(s) field. 4. Press the Save soft key to accept the change.
Configuring Basic Features Phone User Interface Configure the call log. To configure the call log via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Save Call Calllog. 3. Click Confirm to accept the change. To configure the call log via phone user interface: 1. Press Menu->Call Feature->Others->General. 2. Press or , or the Switch soft key to select the desired value from the Save Calllog field. 3.
Administrator’s Guide for SIP-T46G IP Phone Procedure Missed call log can be configured using the configuration files or locally. Configure the missed call log Configuration File .cfg feature. For more information, refer to Missed Call Log on page 254. Configure the missed call log feature. Local Web User Interface Navigate to: http:///servlet ?p=account-basic&q=load&acc =0 To configure missed call log via web user interface: 1. Click on Account. 2.
Configuring Basic Features Procedure Configuration changes can be performed using the configuration files or locally. Specify the access URL of the local contact file. Configuration File .cfg For more information, refer to Access URL of Local Contact File on page 349. Add a new group and a contact to the IP phone.
Administrator’s Guide for SIP-T46G IP Phone To add a contact to the local directory via web user interface: 1. Click on Contacts->Contacts. 2. Enter the name and the office, mobile or other numbers in the corresponding fields. 3. Select the desired ring tone from the pull-down list of Ring Tone. 4. Select the desired group from the pull-down list of Group. 5. Select the desired account from the pull-down list of Account. 6. Select the desired photo from the pull-down list of Photo. 7.
Configuring Basic Features If Auto is selected, the IP phone will use the first available account when placing calls to the contact from the local directory. 6. Press or , or the Switch soft key to select the desired ring tone from the Ring or , or the Switch soft key to select the desired photo from the Photo field. 7. Press field. 8. Press the Save soft key to accept the change.
Administrator’s Guide for SIP-T46G IP Phone 3. (If enabled) Enter the desired delay time (in seconds) in the Inter Digit Time (1~14s) field. 4. Click Confirm to accept the change. The call waiting feature allows the IP phones to receive a new call when there is already an active call. The new call is presented to the user visually on the LCD screen. The call waiting tone feature enables the IP phones to play a short tone when receiving a new incoming call during a conversation.
Configuring Basic Features To configure call waiting via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Call Waiting. 3. (Optional.) Enter the call waiting on code in the Call Waiting On Code field. 4. (Optional.) Enter the call waiting off code in the Call Waiting Off Code field. 5. Click Confirm to accept the change. To configure the call waiting tone via web user interface: 1. Click on Features->Audio. 2.
Administrator’s Guide for SIP-T46G IP Phone To configure call waiting and call waiting tone via phone user interface: 1. Press Menu->Call Feature->Call Waiting. 2. Press or , or the Switch soft key to select the desired value from the Call 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 On Code field. 5. (Optional.) Enter the call waiting off code in the Off Code field. 6.
Configuring Basic Features The default times are 10. 5. Click Confirm to accept the change. To configure auto redial via phone user interface: 1. Press Menu->Call Feature->Others->Auto Redial. 2. Press or , or the Switch soft key to select the desired value from the Auto Redial field. 3. Enter the desired time in the Redial Interval field. 4. Enter the desired times in the Redial Times field. 5. Press the Save soft key to accept the change.
Administrator’s Guide for SIP-T46G IP Phone http:///servlet ?p=account-basic&q=load&acc =0 Phone User Interface Configure the auto answer feature. 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. To configure auto answer via phone user interface: 1.
Configuring Basic Features specified in draft-poetzl-sipping-call-completion-00, to subscribe to the busy party and receive notifications of status changes of the busy party. Procedure Call completion can be configured using the configuration files or locally. Configure the call completion Configuration File .cfg feature. For more information, refer to Call Completion on page 258. Configure the call completion feature.
Administrator’s Guide for SIP-T46G IP Phone The anonymous call feature allows the caller to block the identity from showing up to the callee when placing a call. The callee’s phone LCD screen prompts an incoming call from anonymity. The example of the SIP header for anonymity for reference: Via: SIP/2.0/UDP 10.2.8.183:5063;branch=z9hG4bK1535948896 From: "Anonymous" ;tag=128043702 To: Call-ID: 1773251036@10.2.8.183 CSeq: 1 INVITE Contact:
Configuring Basic Features To configure the 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 Anonymous Call. 5. (Optional.) Enter the anonymous call on code in the On Code field. 6. (Optional.) Enter the anonymous call off code in the Off Code field. 7. Click Confirm to accept the change.
Administrator’s Guide for SIP-T46G IP Phone Procedure Anonymous call rejection can be configured using the configuration files or locally. Configure the anonymous call rejection feature. Configuration File .cfg For more information, refer to Anonymous Call Rejection on page 259. Configure the anonymous call rejection feature. Web User Interface Local Navigate to: http:///servlet ?p=account-basic&q=load&acc =0 Phone User Interface Configure the anonymous call rejection feature.
Configuring Basic Features 2. Select the desired line and then press Enter soft key. 3. Press or , or the Switch soft key to select the desired value from the Anonymous Reject field. 4. (Optional.) Enter the anonymous call rejection on code in the On Code field. 5. (Optional.) Enter the anonymous call rejection off code in the Off Code field. 6. Press the Save soft key to accept the change. The Do Not Disturb (DND) feature allows the IP phones to ignore incoming calls.
Administrator’s Guide for SIP-T46G IP Phone Specify the return code and the reason of the SIP response message. For more information, refer to Do Not Disturb on page 261. Assign a DND key. Navigate to: http:///servlet? p=dsskey&model=1&q=load&line page=1 Configure the DND feature. Navigate to: Web User Interface http:///servlet? p=features-forward&q=load Local Specify the return code and the reason of the SIP response message.
Configuring Basic Features To configure the DND feature via web user interface: 1. Click on Features->Forward & DND. 2. In the DND block, mark the desired radio box in the Mode field. a) If you select Phone: 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 select Custom: 1) Select the desired account from the pull-down list of Account.
Administrator’s Guide for SIP-T46G IP Phone 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 via web user interface: 1. Click on Features->General Information. 2. Select the desired type from the pull-down list of Return Code DND. 3. Click Confirm to accept the change. To configure a DND key via phone user interface: 72 1. Press Menu->Call Feature->DSS Keys. 2. Select the desired DSS key.
Configuring Basic Features 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 Event field. 5. (Optional.) Enter the string that will appear on the LCD screen in the Label field. 6. Press the Save soft key to accept the change. To configure DND in the phone mode via phone user interface: 1. Press the DND soft key or the DND key when the IP phone is idle.
Administrator’s Guide for SIP-T46G IP Phone To configure busy tone delay via web user interface: 1． Click on Features->General Information. 2． Select the desired value from the pull-down list of Busy Tone Delay (Seconds). 3． Click Confirm to accept the change. The return code when refuse feature defines the return code and reason of the SIP response message for call rejection. The caller’s phone LCD screen displays the reason according to the return code received.
Configuring Basic Features http:///servlet ?p=features-general&q=load To configure the return code 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 Refuse. 3. Click Confirm to accept the change. Early media refers to media (e.g., audio and video) played to the caller before a SIP call is actually established. Current implementation supports early media through the 183 message.
Administrator’s Guide for SIP-T46G IP Phone workaround feature. For more information, refer to 180 Ring Workaround on page 265. Configure the 180 ring workaround feature. Local Web User Interface Navigate to: http:///servlet ?p=features-general&q=load To configure 180 ring workaround via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of 180 Ring Workaround. 3. Click Confirm to accept the change.
Configuring Basic Features Procedure Use outbound proxy in dialog can be configured using the configuration files or locally. Specify whether to use outbound proxy in a dialog. Configuration File .cfg For more information, refer to Use Outbound Proxy in Dialog on page 265. Specify whether to use outbound proxy in a dialog.
Administrator’s Guide for SIP-T46G IP Phone message. The re-transmitting and doubling of T1 continues until the retransmitting time reaches the T2 value. Timer T4 represents the time the network will take to clear messages between the SIP client and SIP server. These session timers are configurable on IP phones. Procedure SIP session timer can be configured using the configuration files or locally. Configure the SIP session timer Configuration File .cfg feature.
Configuring Basic Features The default value is 5s. 7. Click Confirm to accept the change. The session timer feature allows for a periodic refresh of SIP sessions through a re-INVITE or an UPDATE request to determine whether the SIP session is still active. Session timer is specified in RFC 4028. The IP phones support two refresher modes: UAC and UAS. The UAC mode means refreshing the session from the client, while the UAS mode means refreshing the session from the server.
Administrator’s Guide for SIP-T46G IP Phone To configure the 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. Select the desired value from the pull-down list of Enable Session Timer. 5. Enter the desired time interval in the Session Expires (Seconds) field. 6. Select the desired refresher from the pull-down list of Session Refresher. 7. Click Confirm to accept the change.
Configuring Basic Features Procedure Call hold can be configured using the configuration files or locally. Configure the call hold tone and call hold tone delay. Specify whether RFC 2543 Configuration File .cfg (c=0.0.0.0) outgoing hold signaling is used. For more information, refer to Call Hold on page 268. Configure the call hold tone and call hold tone delay. Specify whether RFC 2543 Local Web User Interface (c=0.0.0.0) outgoing hold signaling is used.
Administrator’s Guide for SIP-T46G IP Phone 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. 4. Click Confirm to accept the change. The call forward feature allows users to redirect an incoming call to a third party. The IP phones support to redirect an incoming INVITE message by responding with a 302 Moved Temporarily message. This response contains a Contact header with a new URI that should be tried.
Configuring Basic Features Forward International The forward international feature allows users to forward an incoming call to an international telephone number. This feature is disabled by default. Procedure Call forward can be configured using the configuration files or locally. Configure the call forward .cfg feature in custom mode. For more information, refer to Call Forward on page 269. Configure the call forward mode. Configuration File Configure the call forward .
Administrator’s Guide for SIP-T46G IP Phone 2) Enter the destination number you want to forward in the Target field. 3) (Optional.) Enter the on code and off code in the On Code and Off Code fields. 4) Select the ring time to wait before forwarding from the pull-down list of After Ring Times (only for the no answer forward). b) If you select Custom: 1) Select the desired account from the pull-down list of Account. 2) Mark the desired radio box in the Always Forward/Busy Forward/No Answer Forward field.
Configuring Basic Features 4) Select the ring time to wait before forwarding from the pull-down list of After Ring Times (only for the no answer forward). 3. Click Confirm to accept the change. To configure the forward international feature via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Fwd International. 3. Click Confirm to accept the change. To configure call forward in phone mode via phone user interface: 1.
Administrator’s Guide for SIP-T46G IP Phone 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 Forward field. 2) Enter the destination number you want to forward all incoming calls to in the Target field. 3) (Optional.) Enter the always forward on code and off code respectively in the On Code and Off Code fields.
Configuring Basic Features 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 as below: 1) Press or to highlight the Always Forward field. 2) Press the All Lines soft key. The LCD screen prompts ―Copy to All Lines?‖. 3) Press the OK soft key to accept the change.
Administrator’s Guide for SIP-T46G IP Phone Call transfer enables the IP phones to transfer an existing call to another party. The 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. Semi-attended Transfer -- Transfer a call after hearing the ringback tone.
Configuring Basic Features Transfer on Hook and Semi Attended Transfer on Hook. 3. Click Confirm to accept the change. Network conference, also known as centralized conference, provides users with flexibility of call with multiple participants (more than three). IP phones implement network conference using the REFER method specified in RFC 4579. This feature depends on support from a SIP server. Procedure Network conference can be configured using the configuration files or locally.
Administrator’s Guide for SIP-T46G IP Phone 3. Click on Advanced. 4. Select Network 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. For local conference, all parties release the call when the conference initiator drops the conference call. The transfer on conference hang up feature allows the other two parties remain connected when the conference initiator drops the conference call.
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 is used for picking up an incoming call on a specific extension. A user can pick up the incoming call using a directed pickup key or the DPickup soft key.
Administrator’s Guide for SIP-T46G IP Phone Directed Call Pickup on page 282. Assign a directed call pickup key. For more information, refer to Directed Call Pickup Key on page 356. .cfg Configure the directed call pickup feature on a phone basis. For more information, refer to Directed Call Pickup on page 281. Assign a directed call pickup key. Navigate to: http:///servl et?p=dsskey&model=1&q=loa d&linepage=1 Configure the directed call pickup feature on a phone basis.
Configuring Basic Features field. 4. Select the desired line from the pull-down list of Line. 5. Click Confirm to accept the change. To configure the 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-T46G IP Phone 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->Call Feature->DSS 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 Pick Up from the Key Event field. 5.
Configuring Basic Features Procedure Group call pickup can be configured using the configuration files or locally. Configure the group call pickup .cfg code on a per-account basis. For more information, refer to Group Call Pickup on page 283. Assign a group call pickup key. For more information, refer to Configuration File Group Call Pickup Key on page .cfg 357. Configure the group call pickup feature on a phone basis. For more information, refer to Group Call Pickup on page 282.
Administrator’s Guide for SIP-T46G IP Phone 4. Select the desired line from the pull-down list of Line. 5. Click Confirm to accept the change. To configure the 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.
Configuring Basic Features 4. Enter the group call pickup code in the Group Call Pickup Code field. 5. Click Confirm to accept the change. To configure a group pickup key via phone user interface: 1. Press Menu->Call Feature->DSS Keys. 2. Select the desired DSS key. 3. Press or , or the Switch soft key to select Key Event from the Type field. 4.
Administrator’s Guide for SIP-T46G IP Phone The example of the dialog-info message carried in NOTIFY message for reference:
Administrator’s Guide for SIP-T46G IP Phone 2. In the desired DSS key field, select Call Return from the pull-down list of Type. 3. Click Confirm to accept the change. To configure a call return key via phone user interface: 1. Press Menu->Call Feature->DSS 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 Event field. 5. (Optional.
Configuring Basic Features d&linepage=1 Phone User Interface Assign a call park key. To configure a call park key via web user interface: 1. Click on DSSKey->Line Key. 2. In the desired DSS key field, select Call Park from the pull-down list of Type. 3. Enter the desired value (e.g., call park feature code) in the Value field. 4. Select the desired line from the pull-down list of Line. 5. Click Confirm to accept the change. To configure a call park key via phone user interface: 1.
Administrator’s Guide for SIP-T46G IP Phone are configurable. Procedure Web server type can be configured using the configuration files or locally. Specify the web access type, Configuration File .cfg HTTP port and HTTPS port. For more information, refer to Web Server Type on page 284. Specify the web access type, HTTP port and HTTPS port. Local Web User Interface Navigate to: http:///servl et?p=network-adv&q=load Phone User Interface Specify the web access type.
Configuring Basic Features 6. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 7. Click OK to reboot the IP phone. To configure the web server type via phone user interface: 1. Press Menu->Advanced (password: admin) ->Network->Webserver Type. 2. Press or , or the Switch soft key to select the desired value in the HTTP Status field. 3. Enter the HTTP port in the HTTP Port field. 4.
Administrator’s Guide for SIP-T46G IP Phone To configure the presentation of the caller identity 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 the CID Source. 5. Click Confirm to accept the change. The connected line identification presentation (COLP) feature allows IP phones to display the identity of the callee specified for outgoing calls.
Configuring Basic Features 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.
Administrator’s Guide for SIP-T46G IP Phone same VoIP codec as your voice and is audible to the conversation partners. SIP INFO DTMF digits are transmitted by the SIP INFO messages when the voice stream is established after a successful SIP 200 OK-ACK message sequence. The SIP INFO message is sent along the signaling path of the call. The SIP INFO message can support transmitting DTMF digits in three ways: DTMF, DTMF-Relay and Telephone-Event.
Configuring Basic Features If SIP INFO or AUTO+SIP INFO is selected, select the desired value from the pull-down list of DTMF Info Type. 5. Enter the desired value in the DTMF Payload Type (96~255) field. 6. Click Confirm to accept the change. To configure the number of times to send the end RTP Event packet via web user interface: 1. Click on Features->General Information. 2. Select the desired value (1-3) from the pull-down list of DTMF Repetition.
Administrator’s Guide for SIP-T46G IP Phone 3. Click Confirm to accept the change. The suppress DTMF display feature allows the IP phones to suppress the display of DTMF digits. The DTMF digits are displayed as ―*‖ on the phone LCD screen. The suppress DTMF display delay feature defines whether to display the DTMF digits for a short period before displaying ―*‖. Procedure Configuration changes can be performed using the configuration files or locally.
Configuring Basic Features 3. Select the desired value from the pull-down list of Suppress DTMF Display Delay. 4. Click Confirm to accept the change. On some traditional servers, call transfer is implemented via DTMF. The IP phones support to send the specified DTMF digits to the server for transferring a call to a third party. Procedure Configuration changes can be performed using the configuration files or locally. Configure the transfer via DTMF Configuration File .cfg feature.
Administrator’s Guide for SIP-T46G IP Phone 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. The intercom feature allows establishing an audio conversation directly. The called phone picks up intercom calls automatically and establishes intercom conversations. This feature depends on support from a SIP server.
Configuring Basic Features http:///servlet ?p=dsskey&model=1&q=load&li nepage=1 Phone User Interface Assign an intercom key. To configure an intercom key via web user interface: 1. Click on DSSKey->Line Key. 2. In the desired DSS 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.
Administrator’s Guide for SIP-T46G IP Phone Accept Intercom Accept Intercom allows the IP phones to automatically answer an incoming intercom call. Intercom Mute Intercom Mute allows the IP phones to mute the microphone for incoming intercom calls. Warning Tone Warning Tone allows the IP phones to play a warning tone before answering an intercom call. Intercom Barge Intercom Barge allows the IP phones to automatically answer an incoming intercom call while there is already an active call on the IP phone.
Configuring Basic Features 2. Select the desired values from the pull-down lists of Allow 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, Warning Tone and Intercom Barge fields. 3. Press the Save soft key to accept the change.
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Configuring Advanced Features This chapter provides information for making configuration changes for the following advanced features: Distinctive Ring Tones Tones Remote Phonebook LDAP Busy Lamp Field Music on Hold Automatic Call Distribution Message Waiting Indicator Multicast Paging Call Recording Hot Desking Action URL Action URI Server Redundancy LLDP VLAN VPN Quality of Service Network Address Translation SNMP 802.
Administrator’s Guide for SIP-T46G IP Phone appropriate ring tone. The Alert-Info header is in the following two formats: Alert-Info: 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=1,2,3,4,5). Example: Alert-Info: http://127.0.0.1/Bellcore-dr1 The following table identifies the different Bellcore ring tone patterns and cadences.
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 failing to download the file, the IP phone will plays the local ring tone associated with info text. If there is no text matched, the IP phone will play the local ring tone configured on the IP phone in about ten seconds. Example: Alert-Info: http://192.168.0.12:8080/ring.
Administrator’s Guide for SIP-T46G IP Phone 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 Tone. 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. 4. 118 Click Confirm to accept the change.
Configuring Advanced Features When receiving a message or recording a call, the IP phone will play a warning tone. You can customize tones or select the tones customized for a specific country to indicate different conditions of the IP phone. Tone sets vary from country to country. The default tones used on the IP phones are the tone sets of US.
Administrator’s Guide for SIP-T46G IP Phone Configured tones can be heard on the IP phone for the following conditions: Condition Description Dial When in the pre-dialing interface Ring Back Ring-back tone Busy When the callee is busy Congestion When the network is congested Call Waiting Call waiting tone Dial Recall Call hold tone Record When recording a call Info When receiving a special message Stutter When receiving a voice mail Message When receiving a text message Auto Answer W
Configuring Advanced Features If you select Custom, you can customize the tone for indicating each condition of the IP phone. 3. Click Confirm to accept the change. Remote phonebook is the phone book maintained centrally, which is stored on the remote server. Users just need the access URL of the remote phonebook. The IP phone can establish a connection with the remote server and download the entries, and then display the entries on the phone user interface.
Administrator’s Guide for SIP-T46G IP Phone phonebook when the IP phone receives incoming calls. Specify how often the IP phone refreshes the local cache of the remote phonebook. For more information, refer to Remote Phonebook on page 296. Specify the access URL of the remote phonebook. Navigate to: http:///servl et?p=contacts-remote&q=load Specify whether to query the contact names from the remote Local Web User Interface phonebook when the IP phone receives incoming calls.
Configuring Advanced Features To configure the remote phonebook via web user interface: 1. Click on Contacts->Remote Phone Book. 2. Select the desired value from the pull-down list of SRemote Name. 3. Enter the desired time in the SRemote Name Flash Time (Seconds) field. 4. Click Confirm to accept the change. LDAP (Lightweight Directory Access Protocol) is an application protocol for accessing and maintaining information services of the distributed directory over an IP network.
Administrator’s Guide for SIP-T46G IP Phone LDAP Attributes The following table lists the most common attributes used to configure the LDAP lookup on IP phones: Abbreviation Name gn givenName Description First name LDAP attribute being made up cn commonName from given name joined to surname.
Configuring Advanced Features To configure LDAP via web user interface: 1. Click on Contacts->LDAP. 2. Select Enabled from the pull-down list of Enable LDAP. 3. Enter the values in the corresponding fields. 4. Select the desired values from the corresponding pull-down lists. 5. Click Confirm to accept the change. To configure an LDAP key via web user interface: 1. Click on DSSKey->Line Key. 2. In the desired DSS key field, select LDAP from the pull-down list of Type. 3.
Administrator’s Guide for SIP-T46G IP Phone To configure an LDAP key via phone user interface: 1. Press Menu->Call Feature->DSS 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 LDAP from the Key Event field. 5. (Optional.) Enter the string that will appear on the LCD screen in the Label field. 6. Press the Save soft key to accept the change.
Configuring Advanced Features Line key LED (configured as BLF key when LED Off in Idle is enabled) LED Status Description The monitored user is busy. Solid red The call is parked against the monitored user’s phone number. Fast flashing red Off The monitored user receives an incoming call. The monitored user is idle. The monitored user does not exist. Procedure BLF can be configured using the configuration files or locally. Assign a BLF key. For more information, refer to BLF Key on page 360.
Administrator’s Guide for SIP-T46G IP Phone Phone User Interface Assign a BLF key. To configure a BLF key via web user interface: 1. Click on DSSKey->Line Key. 2. In the desired DSS key field, select BLF from the pull-down list of Type. 3. Enter the phone number or extension you want to monitor in the Value field. 4. Select the desired line from the pull-down list of Line. 5. (Optional.) Enter the directed call pickup code in the Extension field. 6. Click Confirm to accept the change.
Configuring Advanced Features To configure the 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. To configure a BLF key via phone user interface: 1. Press Menu->Call Feature->DSS Keys. 2. Select the desired DSS key. 3. Press or , or the Switch soft key to select BLF from the Type field. 4.
Administrator’s Guide for SIP-T46G IP Phone Internet) to the held party. Procedure Music on Hold can be configured using the configuration files or locally. Configure the MoH feature on a Configuration File .cfg per-account basis. For more information, refer to Music on Hold on page 303. Configure the MoH feature on a per-account basis. Local Web User Interface Navigate to: http:///servlet ?p=account-adv&q=load&acc= 0 To configure the MoH feature via web user interface: 130 1.
Configuring Advanced Features Automatic Call Distribution (ACD) enables organizations to manage a large number of phone calls on an individual basis. ACD enables the use of the IP phones in a call-center role by automatically distributing incoming calls to available users, or agents. The ACD feature depends on support from a SIP server. Note The ACD feature is disabled by default. You need to enable it in advance.
Administrator’s Guide for SIP-T46G IP Phone ?p=features-acd&q=load Phone User Interface Assign an ACD key. To configure an ACD key via web user interface: 1. Click on DSSKey->Line Key. 2. In the desired DSS key field, select ACD from the pull-down list of Type. 3. Select the desired line from the pull-down list of Line. 4. Click Confirm to accept the change. To configure the ACD auto available timer feature via web user interface: 132 1. Click on Features->ACD. 2.
Configuring Advanced Features To configure an ACD key via phone user interface: 1. Press Menu->Call Feature->DSS Keys. 2. Select the desired DSS key. 3. Press 4. (Optional.) Enter the string that will appear on the LCD screen in the Label field. 5. Press the Save soft key to accept the change. or , or the Switch soft key to select ACD from the Type field. Message Waiting Indicator (MWI) is a feature that informs users that they have messages waiting in their mailboxes.
Administrator’s Guide for SIP-T46G IP Phone To configure the MWI subscription feature via web user interface: 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3. Click on Advanced. 4. Select the desired value from the pull-down list of Subscribe for MWI. 5. Enter the period time in the MWI Subscription Period (Seconds) field. 6. Click Confirm to accept the change. The IP phone will subscribe to the account number for MWI service by default.
Configuring Advanced Features 5. Enter the desired voice number in the Voice Mail field. 6. Click Confirm to accept the change. The IP phone will subscribe to the voice mail number for MWI service using Subscribe MWI to WM feature. The multicast paging feature allows the IP phones to send/receive Real-time Transport Protocol (RTP) stream to/from the pre-configured multicast address(es) without involving SIP signaling. You can specify up to 10 listening multicast addresses on IP phones.
Administrator’s Guide for SIP-T46G IP Phone 362. Specifies a multicast codec for the IP phone to use for multicast RTP. For more information, refer to Sending RTP Stream on page 307. Assign a multicast paging key. Navigate to: http:///servlet ?p=dsskey&model=1&q=load&li nepage=1 Local Web User Interface Specifies a multicast codec for the IP phone to use to send the RTP stream.
Configuring Advanced Features 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->Call Feature->DSS 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 Paging from the Key Event field. 5. (Optional.
Administrator’s Guide for SIP-T46G IP Phone 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 on the IP phone. If the parameter is configured as disabled, the IP phone will automatically ignore all incoming multicast paging calls.
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 the paging barge and paging priority active features via web user interface: 1. Click on Contacts->MulticastIP. 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-T46G IP Phone Call recording enables users to record calls. It depends on support from a SIP server. When the user presses the call record key, the IP phone sends a record request to the server. The IP phones themselves do not have memory to store the recording, what they can do is to trigger the recording and indicate the recording status. Normally, there are 2 main methods to trigger a recording on a certain server. We call them record and URL record.
Configuring Advanced Features Content-Length: 0 URL Record When a user presses a URL record key for the first time during a call, the IP phone sends an HTTP GET message to the server. The example of an HTTP GET message for reference: Get /phonerecording.cgi?model=yealink HTTP/1.0\r\n Request Method: GET Request URI: /phonerecording.cgi?model=yealink Request version: HTTP/1.0 Host: 10.1.2.224\r\n User-agent: yealink SIP-T46G 28.71.0.
Administrator’s Guide for SIP-T46G IP Phone The recording session is successfully stopped. 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 363. Assign a URL record key. For more information, refer to URL Record Key on page 363. Assign a record key. Assign a URL record key.
Configuring Advanced Features 2. In the desired DSS key field, select URL Record from the pull-down list of Type. 3. Enter the URL in the Value field. 4. Click Confirm to accept the change. To configure a record key via phone user interface: 1. Press Menu->Call Feature->DSS 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 Event field. 5. (Optional.
Administrator’s Guide for SIP-T46G IP Phone The hot desking feature allows a user to delete all accounts on the IP phone, register his account on line 1. In order to use this feature, you need to assign a hot desking key. Procedure 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 364. Assign a hot desking key.
Configuring Advanced Features Action URL allows IP phones to interact with web server applications by sending an HTTP or HTTPS GET request. You can specify a URL that triggers a GET request when a specified event occurs. Action URL can be only triggered by the pre-defined events (e.g., log on). The valid URL formats are: http://IP address of the server/help.xml? and https://IP address of the server/help.xml? The following table lists the pre-defined events for action URL.
Administrator’s Guide for SIP-T46G IP Phone Event Description Mute When the IP phone mutes a call. Unmute When the IP phone unmutes a call. Missed Call When the IP phone misses a call. IP Changed When the IP address of the IP phone changes. 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.
Configuring Advanced Features Variable Value Description call or establishes a call. The SIP URI of the caller when the IP phone places a $local 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.
Administrator’s Guide for SIP-T46G IP Phone 2. Enter the action URLs in the corresponding fields. 3. Click Confirm to accept the change. Opposite to action URL, action URI allows IP phones to interact with web server application by receiving and handling an HTTP or HTTPS GET request. When receiving a GET request, the IP phone will perform the specified action and respond with a 200 OK message. A GET request may contain variable named as ―key‖ and variable value, which are separated by ―=‖.
Configuring Advanced Features Variable Value Note Phone Action L1-L27 Press the Line key. F_CONFERENCE Press the Conference soft key. F1-F4 Press the soft key. MSG Press the MESSAGE key. HEADSET Press the HEADSET key. RD Press the REDIAL key. UP/DOWN/LEFT/RIGHT Press the Navigation keys. Reboot Reboot the IP phone. AutoP Let the IP phone perform auto provisioning. DNDOn Activate the DND mode. DNDOff Deactivate the DND mode. The variable value does not work with all events.
Administrator’s Guide for SIP-T46G IP Phone 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. If you enter ―any‖ in this field, the IP phone can receive and handle GET requests from any IP address. If you leave the field blank, the IP phone cannot receive or handle any HTTP GET request. 3.
Configuring Advanced Features Phone Configuration for Redundancy Implementation To assist in explaining the redundancy behavior, an illustrative example of how an IP phone may be configured is shown next. In the example, server redundancy for fallback and fail-over purposes is deployed. Two separate SIP servers (a working server 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, sip:user@example.
Administrator’s Guide for SIP-T46G IP Phone When registering to the working server, the IP phone must always register to the primary server first except in failover conditions. When the primary server registration is unavailable, the secondary server will serve as the working server. 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 discovered through DNS as specified by RFC 3263.
Configuring Advanced Features Parameter Description replacement Specify a domain name to be used for the next query. The IP phone picks the first record, because its order of 90 is lower than 100. The pref parameter is unimportant as there is no other record with order 90. The flag ―s‖ indicates performing the SRV query next. TCP will be used, targeted to a host determined by an SRV query of ―_sip._tcp.example.com‖.
Administrator’s Guide for SIP-T46G IP Phone the call: 1. Send the INVITE request to the primary server. 2. If the primary server does not respond correctly to the INVITE, then try and make the call using the secondary server. 3. If the secondary server is also unavailable, the IP phone will try the fallback server until it either succeeds in making a call or exhausts all servers at which point the call will fail. At the start of a call, server availability is determined by SIP signaling failure.
Configuring Advanced Features 5. Configure parameters of the SIP server 2 in the corresponding fields. 6. Click Confirm to accept the change. LLDP (Linker Layer Discovery Protocol) is a vendor-neutral Link Layer protocol. It allows IP phones to receive and/or transmit device-related information to directly connected devices on the network that are also using the protocol, and store the information that is learned about other devices.
Administrator’s Guide for SIP-T46G IP Phone Power Management -- provides information related to how the IP phones are powered, power priority, and how much power IP phones need. Inventory Management -- provides a means to effectively manage the IP phones and the attributes of the IP phones such as model number, serial number and software revision.
Configuring Advanced Features TLV Type TLV Name Description The supported LLDP-MED TLV types are: LLDP-MED Capabilities, Network Policy, Extended Power via MDI-PD, Inventory. Network Policy Extended Power-via-MDI Inventory – Hardware Revision Inventory – Firmware Revision Inventory – Software Revision Inventory – Serial Number Port VLAN ID, application type, L2 priority and DSCP value. Power type, source, priority and value. Hardware revision of phone. Firmware revision of phone.
Administrator’s Guide for SIP-T46G IP Phone The valid values range from 1 to 3600. 4. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 5. Click OK to reboot the IP phone. 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. For more information, refer to VLAN on page 317.
Administrator’s Guide for SIP-T46G IP Phone 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 the 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 the settings will take effect after reboot. 5. Click OK to reboot the IP phone.
Administrator’s Guide for SIP-T46G IP Phone more prevalent due to the benefits: scalability, reliability, convenience and security. There are two types of VPN access: remote-access VPN (connecting an individual device to a network) and site-to-site VPN (connecting two networks together). Remote-access VPN allows employees to access their company's intranet from home or outside the office, and site-to-site VPN allows employees in geographically separated offices to share one cohesive virtual network.
Configuring Advanced Features 2. Click Browse to locate the tar package 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 VPN Active. 5. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 6. Click OK to reboot the IP phone.
Administrator’s Guide for SIP-T46G IP Phone QoS provides better network service by providing the following features: Supporting dedicated bandwidth Improving loss characteristics Avoiding and managing network congestion Shaping network traffic Setting traffic priorities across the network The Best-Effort service is the default QoS model in the IP networks. It provides no guarantees for data delivering, which means delay, jitter, packet loss and bandwidth allocation are unpredictable.
Configuring Advanced Features Voice QoS For VoIP transmissions to be intelligible to the receiver, voice packets should not be dropped, excessively delayed, or suffer varying delay. DiffServ model can guarantee high-quality voice transmission when the voice packets are configured higher DSCP value. SIP QoS SIP protocol is used for creating, modifying and terminating two-party or multi-party sessions.
Administrator’s Guide for SIP-T46G IP Phone 3. Enter the desired value in the SIP QoS (0~63) field. 4. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 5. Click OK to reboot the IP phone. Network Address Translation (NAT) is essentially a translation table that maps public IP address and port combinations to private IP address and port combinations. This reduces the need for a large amount of public IP addresses.
Configuring Advanced Features assistance from a third-party network server (STUN server) usually located on public Internet. The IP phone can be configured to act as a STUN client, which sends exploratory STUN messages to the STUN server. The STUN server uses those messages to determine the public IP address and port used, and then informs the client. The NAT traversal and STUN server are configurable on a per-account basis.
Administrator’s Guide for SIP-T46G IP Phone 5. Click Confirm to accept the change. SNMP (Simple Network Management Protocol) is an Internet-standard protocol for managing devices on IP networks. It is used mostly in network management systems to monitor network-attached devices for conditions that warrant administrative attention. SNMP exposes management data in the form of variables on the managed systems, which describe the system configuration.
Configuring Advanced Features Procedure SNMP can be configured using the configuration files or locally. Configure SNMP on the IP Configuration File .cfg phone. For more information, refer to SNMP on page 322. Configure SNMP. Local Web User Interface Navigate to: http:///servl et?p=network-adv&q=load To configure SNMP via web user interface: 1. Click on Network->Advanced. 2. In the SNMP block, select the desired value from the pull-down list of Active. 3.
Administrator’s Guide for SIP-T46G IP Phone IEEE 802.1X authentication is an IEEE standard for Port-based Network Access Control (PNAC). It is part of the IEEE 802.1 group of networking protocols. It provides an authentication mechanism to devices wishing to attach to a LAN or WLAN. The 802.1X authentication involves three parties: a supplicant, an authenticator and an authentication server. The supplicant is the IP phone that wishes to attach to the LAN or WLAN. With 802.
Configuring Advanced Features 2) Enter the password for authentication in the MD5 Password field. b) If you select EAP-TLS: 1) Enter the username for authentication in the Identity field. 2) Leave the MD5 Password field blank. 3) In the CA Certificate field, click Browse to select the desired CA certificate (*.pem,*.crt, *.cer or *.der) from your local system. 4) In the Device Certificate field, click Browse to select the desired client certificate (*.pem or *.cer) from your local system.
Administrator’s Guide for SIP-T46G IP Phone 5) Click Upload to upload the certificates. c) If you select PEAP-MSCHAPV2: 1) Enter the username for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificate field, click Browse to select the desired certificate (*.pem,*.crt, *.cer or *.der) from your local system. 4) Click Upload to upload the certificate.
Configuring Advanced Features 1) Enter the username for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificate field, click Browse to select the desired certificate (*.pem,*.crt, *.cer or *.der) from your local system. 4) Click Upload to upload the certificate. 3. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 4. Click OK to reboot the IP phone.
Administrator’s Guide for SIP-T46G IP Phone 1) Enter the username for authentication in the Identity field. 2) Enter the password for authentication in the Password field. 3. Click Save to accept the change. The IP phone reboots automatically to make the settings effective after a period of time. TR-069 is a technical specification, which is defined by the Broadband Forum.
Configuring Advanced Features RPC Method Description specified file from the designated location. 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.
Administrator’s Guide for SIP-T46G IP Phone and ACS Password fields. 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 username and password authenticated by the IP phone in the Connection Request Username and Connection Request Password fields. 8. Click Confirm to accept the change.
Configuring Advanced Features router is configured by the network administrator and sends out Router Advertisement announcements onto the link. These announcements can allow the on-link connected IP phone to configure itself with IPv6 address, as specified in RFC 4862. Stateful DHCPv6: The Dynamic Host Configuration Protocol for IPv6 (DHCPv6) has been standardized by the IETF through RFC3315. DHCPv6 enables DHCP servers to pass configuration parameters such as IPv6 network addresses to IPv6 nodes.
Administrator’s Guide for SIP-T46G IP Phone If you mark the Static IP Address radio box, configure the IPv6 address and other configuration parameters in the corresponding fields. 4. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 5. Click OK to reboot the IP phone. To configure the SLAAC feature via web user interface: 1. 178 Click on Network->Advanced.
Configuring Advanced Features 2. In the ICMPv6 Status block, select the desired value from the pull-down list of Active. 3. Click Confirm to accept the change. To configure IPv6 address via phone user interface: 1. Press Menu->Advanced (password: admin) ->Network->WAN Port. 2. Press or to select the desired address mode 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.
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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 The headset prior feature allows users to use headset preferentially if a headset is physically connected to the IP phone. This feature is especially useful for permanent or full-time headset users. Procedure Headset prior can be configured using the configuration files or locally.
Administrator’s Guide for SIP-T46G IP Phone 2. Select the desired value from the pull-down list of Headset Prior. 3. Click Confirm to accept the change. The dual headset feature allows users to use two headsets on one IP phone. To use this feature, the users need to physically connect two headsets to the headset jack and handset jack respectively.
Configuring Audio Features To configure dual headset via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Dual-Headset. 3. Click Confirm to accept the change. CODEC is an abbreviation of COmpress-DECompress. It is capable of coding or decoding a digital data stream or signal by implementing an algorithm.
Administrator’s Guide for SIP-T46G IP Phone codecs. The attribute ―rtpmap‖ is used to define a mapping from RTP payload codes to a codec, clock rate and other encoding parameters.
Configuring Audio Features Packetization Time Ptime (Packetization Time) is measurement of the duration (in milliseconds) of the audio data in each RTP packet sent to the destination, and hence it defines how much network bandwidth is used for transfer of the RTP stream. Before establishing a conversation, codec and ptime are negotiated through SIP signaling. The valid values of ptime range from 10 to 60, in increments of 10 milliseconds. The default ptime is 20ms.
Administrator’s Guide for SIP-T46G IP Phone 7. To adjust the order of the enabled codecs, click 8. Click Confirm to accept the change. or . To configure the Ptime on a per-account basis via web user interface: 186 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3. Click on Advanced. 4. Select the desired value from the pull-down list of PTime (ms). 5. Click Confirm to accept the change.
Configuring Audio Features Acoustic echo cancellation (AEC) is used to remove acoustic echo from a voice communication in order to improve the voice quality. It also increases the capacity achieved through silence suppression by preventing echo from traveling across a network. IP phones employ advanced AEC for hands-free operation. Echo cancellation is done using the echo canceller. Procedure AEC can be configured using the configuration files or locally. Configure the AEC feature.
Administrator’s Guide for SIP-T46G IP Phone Voice Activity Detection (VAD) is used in speech processing to detect the presence or absence of human speech. When detecting period of ―silence‖, VAD replaces that silence efficiently with special packets that indicate silence is occurring. It can facilitate speech processing, and can also be used to deactivate some processes during non-speech section of an audio session.
Configuring Audio Features Comfort Noise Generation (CNG) is used to generate background noise for voice communications during periods of silence that occur during the conversation. It is part of the silence suppression or VAD handling for VoIP technology. CNG, in conjunction with VAD algorithms, quickly determines when periods of silence occur and inserts artificial noise until voice activity resumes.
Administrator’s Guide for SIP-T46G IP Phone Jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in evenly spaced intervals. Jitter is variations in packet arrival time, can occur because of network congestion, timing drift or route changes.
Configuring Audio Features 5. Enter the fixed delay time for fixed jitter buffer in the Normal field. 6. Click Confirm to accept the change.
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Configuring Security Features This chapter provides information for making configuration changes for the following security-related features: Transport Layer Security Secure Real-Time Transport Protocol Encrypting Configuration Files The TLS protocol is a commonly-used protocol for providing communications privacy and managing the security of message transmission.
Administrator’s Guide for SIP-T46G IP Phone The following figure illustrates the TLS messages exchanged between the IP phone and TLS server to establish an encrypted communication channel: Step1: The 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.
Configuring Security Features not overwritten by the new one. The format of the certificates must be *.pem and *.cer. You can specify the IP phone whether to authenticate the certificate sent by the connecting server based on the trusted certificates list. The trusted certificates list and the server certificates list contain the default and custom certificates. You can specify the IP phone to accept the type of certificates: default certificates, custom certificates, or all certificates.
Administrator’s Guide for SIP-T46G IP Phone Navigate to: http:///servl et?p=server-cert&q=load To configure TLS on a per-account basis via web user interface: 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3. Select TLS from the pull-down list of the Transport. 4. Click Confirm to accept the change. To configure the trusted certificates feature via web user interface: 196 1. Click on Security->Trusted Certificates. 2.
Configuring Security Features 4. Select the desired value from the pull-down list of CA Certificates. 5. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 6. Click OK to reboot the IP phone. To upload a trusted certificate via web user interface: 1. Click on Security->Trusted Certificates.
Administrator’s Guide for SIP-T46G IP Phone 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 feature via web user interface: 1. Click on Security->Server Certificates. 2. Select the desired value from the pull-down list of Device Certificates. 3. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 4.
Configuring Security Features 2. Click Browse to select the certificate (*.pem or *.cer) from your local system. 3. Click Upload to upload the certificate. The dialog box pops up to prompt ―Success: The Server Certificate has been loaded! Rebooting, please wait…‖. Secure Real-Time Transport Protocol (SRTP) provides means of encrypting the RTP streams during VoIP phone calls to avoid interception and eavesdropping. The parties participating in the call should enable the SRTP feature simultaneously.
Administrator’s Guide for SIP-T46G IP Phone a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv The callee receives the INVITE message with the RTP encryption algorithm. The callee answers the call and responses with a 200 OK message carrying the negotiated RTP encryption algorithm.
Configuring Security Features To configure the SRTP feature via web user interface: 1. Click on Account. 2. Select the desired account from the pull-down list of Account. 3. Click on Advanced. 4. Select the desired value from the pull-down list of RTP Encryption (SRTP). 5. Click Confirm to accept the change. The IP phone can download the encrypted configuration files from the provisioning server to protect against unauthorized access and tampering of sensitive information (i.e.
Administrator’s Guide for SIP-T46G IP Phone MAC-Oriented AES key is used to encrypt and decrypt the .cfg file. The AES keys must be 16 characters. The AES key should be configured on the IP phone for decrypting before provisioning. Procedure to Encrypt Configuration Files To encrypt the .cfg file: 1. Place the ―EncryptUtilityWindows.exe‖ tool and .cfg file to the same directory (i.e., D:\). 2. Open a command line window application (i.e., DOS window). 3.
Configuring Security Features 2. Enter the values in the Common AES Key and MAC-Oriented AES Key fields. 3. Click Confirm to accept the change.
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Upgrading the Firmware This chapter provides information about upgrading the IP phone firmware. There are two methods used to upgrade the firmware on the IP phone: Upgrade the firmware manually from the local system Upgrade the firmware from the provisioning server automatically. The associated firmware for SIP-T46G IP phone is 28.x.x.x.rom. Note You can download the latest firmware at: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
Administrator’s Guide for SIP-T46G IP Phone 5. Note Click OK to confirm the upgrading. Do not unplug the network and power cables when the IP phone is upgrading the firmware. Do not close the browser when the IP phone is upgrading the firmware via web user interface. Upgrade Firmware from the Provisioning Server IP phones support to use the FTP, TFTP, HTTP, and HTTPS protocols to download the configuration files and firmware from the provisioning server, and then upgrade the firmware automatically.
Upgrading the Firmware 2. Mark the desired radio box in the Power On field. 3. Click Confirm to accept the change. When the ―Power On‖ is set to On, the IP phone will check for both firmware and configuration files stored on the provisioning server during booting up.
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Resource Files When configuring some features, you may need to upload resource files to the IP phone. The resources files can be local contact directory, remote phonebook and so on. Ask the Yealink field application engineer for the resource file templates. If the resource file is to be used for all IP phones of the same model, the access URL of the resource file is best specified in the .cfg file.
Administrator’s Guide for SIP-T46G IP Phone Procedure Use the following procedures to customize a replace rule template. Customizing 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. Customizing 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 leaving it blank, the IP phone will apply to all lines. 3. Specify the values within double quotes. 4.
Administrator’s Guide for SIP-T46G IP Phone indicates the start of the default soft key list and indicates the end of the default soft key list, the default soft keys are displayed on the phone LCD screen by default. Procedure Use the following procedures to customize a softkey layout template. Customizing a softkey layout template: 1. Open the template file using an ASCII editor. 2. For each soft key that you want to enable, add the following string to the file.
Resource Files You can add contacts one by one on the IP phone directly. In some cases, you may want to add multiple contacts to the IP phone at the same time or share the contacts on many IP phones. You can create a local contact file, and then place the local contact file to the root directory of the provisioning server, specify the access URL of the contact file in the configuration files.
Administrator’s Guide for SIP-T46G IP Phone Where: display_name=‖‖ specifies the name of the group. ring=‖‖ specifies the desired ring tone for this group. 4. Specify the values within double quotes. 5. Place this file to the root directory of the provisioning server.
Resource Files 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 root directory of the provisioning server.
Administrator’s Guide for SIP-T46G IP Phone 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 346. 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 349. Configure the access URL of the remote XML phonebook. Configuration File .
Troubleshooting This chapter provides an administrator with general information for troubleshooting some common problems that may encounter while using the SIP-T46G IP phone. The IP phone can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which helps an administrator quickly find out the reasons for the failure and do the troubleshooting more easily.
Administrator’s Guide for SIP-T46G IP Phone 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?‖ 4. Click OK to reboot the IP phone. To export log files to a syslog server via web user interface: 1. Click on Settings->Configuration. 2. In the Export System Log block, mark the Server radio box. 3. Enter the address of the syslog server in the Server Name field. 4.
Troubleshooting system. The following figure shows a portion of a log file: You can capture packets in two ways: capturing the packets via web user interface or using the Ethernet software. You can analyze the packets captured for troubleshooting purposes. To capture packets via web user interface: 1. Click on Settings->Upgrade.
Administrator’s Guide for SIP-T46G IP Phone 2. Click Start to begin capturing signal traffic. 3. Reproduce the issue to get stack traces. 4. Click Stop to end capturing. 5. Click Export to open file download window, and then save the file to your local system. To capture packets using the Ethernet software: Connect the IP phone’s Internet port with the PC to the same HUB, and then use Sniffer, Ethereal or Wireshark software to capture the packets.
Troubleshooting 2. Select the desired value from the pull-down list of WatchDog. 3. Click Confirm to accept the change. Status indicators may consist of the power LED, message key indicator, line key indicator, headset key indicator and the on-screen icon or error messages.
Administrator’s Guide for SIP-T46G IP Phone 2. In the Export or Import Configuration block, click Export to open the file download window, and then save the file to your local system. This section describes solutions to some common scenarios that may occur while using the IP phone. If you encounter a scenario which is not listed in this section, contact your Yealink reseller for further support. Do one of the following: Check that the power LED is on to ensure the IP phone is powered on.
Troubleshooting Ensure that the switch or hub in your network is operational. Press the OK key when the IP phone is idle to check the basic information of the IP phone, such as IP address and firmware version. Do one of the following: Ensure that the target firmware is not the same as the current used firmware. Ensure that the target firmware is applicable to the IP phone model. Ensure that the current or the target firmware is not protected.
Administrator’s Guide for SIP-T46G IP Phone to see if another line provides better connection. A remote phonebook is placed on a server, while a local phonebook is placed on the IP phone flash. A remote phonebook can be used by everyone that can access the server, while a local phonebook can only be used by a specific phone itself. A remote phonebook is always used as a central phonebook for a company.
Troubleshooting The volumes in different cases are separated. You can use the volume key under the navigation keys to increase or decrease the voice volume. You can press the volume key to adjust the ringer volume when the phone is idle. You can also press the volume key to adjust the receiver volume of currently used audio devices (handset, speakerphone or headset), when the phone is in the dialing interface or during a call.
Administrator’s Guide for SIP-T46G IP Phone the server simultaneously. Then the server configures the Always Forward feature as configured on the phone side. Hence, the server is able to get the right status of the extension. Do one of the following: Try to set another available IP address for the IP phone. Check the configuration of the network via phone user interface at the path Menu->Advanced->Network->WAN Port. If Static IP Client is selected, select DHCP IP Client instead.
Appendix 802.1x — an IEEE Standard for port-based Network Access Control (PNAC). It is part of the IEEE 802.1 group of networking protocols. It provides an authentication mechanism to devices wishing to attach to a LAN or WLAN. ACD (Automatic Call Distribution) — used to distribute calls from large volumes of incoming calls to the registered IP phone users. ACS (Auto Configuration server) — responsible for auto-configuration of the Central Processing Element (CPE).
Administrator’s Guide for SIP-T46G IP Phone IEEE (Institute of Electrical and Electronics Engineers) — a non-profit professional association headquartered in New York City that is dedicated to advancing technological innovation and excellence. LAN (Local Area Network) — used to interconnects network devices in a limited area such as a home, school, computer laboratory, or office building. MIB (Management Information Base) — a virtual database used for managing the entities in a communications network.
Appendix Time Zone Time Zone Name −11:00 Samoa −10:00 United States-Hawaii-Aleutian −10:00 United States-Alaska-Aleutian −09:00 United States-Alaska Time −08:00 Canada(Vancouver, Whitehorse) −08:00 Mexico(Tijuana, Mexicali) −08:00 United States-Pacific Time −07:00 Canada(Edmonton, Calgary) −07:00 Mexico(Mazatlan, Chihuahua) −07:00 United States-Mountain Time −07:00 United States-MST no DST −06:00 Canada-Manitoba(Winnipeg) −06:00 Chile(Easter Islands) −06:00 Mexico(Mexico City,
Administrator’s Guide for SIP-T46G IP Phone Time Zone 230 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-T46G IP Phone This appendix describes the parameters you can set in the configuration files for the IP phone. The configuration files are .cfg and .cfg. You can set specific parameters in the configuration files for configuring IP phones. The .cfg and .cfg files are stored on the provisioning server. The IP phone checks for configuration files and looks for resource files when restarting the IP phone. The .
Appendix Parameter- Configuration File network.internet_port.type .cfg Defines 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.internet_port.ip .
Administrator’s Guide for SIP-T46G IP Phone Format IP Address Default Value Blank Range Not Applicable Example network.internet_port.mask = 255.255.255.0 Parameter- Configuration File network.internet_port.gateway .cfg Configures the default gateway when the Internet port type is configured as Static IP Description Address. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Appendix Parameter- Configuration File network.secondary_dns .cfg Configures the secondary DNS server when the Internet port type is configured as Static IP Description Address. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value 202.101.103.56 Range Not Applicable Example network.secondary_dns = 202.101.103.6 Parameter- Configuration File network.internet_port.type .
Administrator’s Guide for SIP-T46G IP Phone Format String Default Value Blank Range Not Applicable Example network.pppoe.user = xmyealink Parameter- Configuration File network.pppoe.password .cfg Configures the PPPoE password when the Internet port type is configured as PPPoE. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format String Default Value Blank Range Not Applicable Example network.pppoe.
Appendix Example network.internet_port.speed_duplex = 0 PC Port Negotiation Parameter- Configuration File network.pc_port.speed_duplex .cfg Specifies the transmission method of PC port. Description Note: We recommend that you do not change this parameter. Format Integer Default Value 0 Valid values are: 0-Auto negotiate 1-Full duplex, 10Mbps Range 2-Full duplex, 100Mbps 3-Half duplex, 10Mbps 4-Half duplex, 100Mbps 5-Full duplex, 1000Mbps Example network.pc_port.
Administrator’s Guide for SIP-T46G IP Phone X ranges from 1 to 20. Format String Default Value Blank Range Not Applicable Example dialplan.replace.replace.1 = 91$12 Parameter- Configuration File dialplan.replace.line_id.X = .cfg Specifies the desired line to apply this replace rule. Description X ranges from 1 to 20. Note: Multiple line IDs are separated by comma. Format String Default Value Blank Range Not Applicable Example dialplan.replace.line_id.
Appendix dial-now rule. X ranges from 1 to 20. Note: Multiple line IDs are separated by comma. Format String Default Value Blank Range Not Applicable Example dialplan.dialnow.line_id.1 = 1,2,3 Parameter- Configuration File phone_setting.dialnow_delay .cfg Configures the delay time (in seconds) for the dial-now rule. Description When entered numbers match the predefined dial-now rule, the IP phone will automatically dial out the entered number after the specified delay time.
Administrator’s Guide for SIP-T46G IP Phone Parameter- Configuration File dialplan.area_code.min_len .cfg Description Sets the minimum length of the entered numbers. Format Integer Default Value 1 Range 1 to 15 Example dialplan.area_code.min_len = 2 Parameter- Configuration File dialplan.area_code.max_len .cfg Sets the maximum length of the entered Description numbers. Note: The value must be larger than the minimum length.
Appendix Block Out Parameter- Configuration File dialplan.block_out.number.x .cfg Description Specifies the block out numbers. X ranges from 1 to 10. Format String Default Value Blank Range Not Applicable Example dialplan.block_out.number.1 = 0000 Parameter- Configuration File dialplan.block_out.line_id.x .cfg Specifies the desired line to apply this block out rule. Description X ranges from 1 to 10. Note: Multiple line IDs are separated by comma.
Administrator’s Guide for SIP-T46G IP Phone Example phone_setting.active_backlight_level = 1 Parameter- Configuration File phone_setting.backlight_time .cfg Configures the backlight time (in seconds) used to specify the delay time to turn off the Description backlight when the IP phone is inactive. If set to 60 (60s), the LCD backlight is turned off when the IP phone is inactive for 60 seconds.
Appendix Parameter- Configuration File security.user_password .cfg Sets a new administrator password for the IP phone. Description The IP phone uses ―admin‖ as the default administrator password. Note: IP phones support ASCII characters 32-126(0x20-0x7E) only in passwords. Format administrator username:new password Default Value admin Range ASCII characters 32-126(0x20-0x7E) Example security.user_password = admin:password000 Parameter- Configuration File phone_setting.
Administrator’s Guide for SIP-T46G IP Phone Parameter- Configuration File phone_setting.phone_lock.unlo .cfg ck_pin Sets a new unlock password. Once the IP phone is locked, you can use ―123‖ as the Description default password to unlock it. Note: IP phones support numeric characters only in password. Format Numeric characters only Default Value 123 Range 0 to 15 characters Example phone_setting.phone_lock.unlock_pin = 123456 Parameter- Configuration File phone_setting.
Appendix NTP Server Parameter- Configuration File local_time.ntp_server1 .cfg Description Sets the IP address or the domain name of the primary NTP server. Format IP Address or Domain Name Default Value cn.pool.ntp.org Range Not Applicable Example local_time.ntp_server1 = 192.168.0.5 Parameter- Configuration File local_time.ntp_server2 .cfg Sets the IP address or the domain name of the secondary NTP server.
Administrator’s Guide for SIP-T46G IP Phone Time Zone Parameter- Configuration File local_time.time_zone .cfg Defines the time zone. Description For more available time zone list, refer to Appendix B: Time Zones on page 229. Format Not Applicable Default Value +8 Range -11 to +13 Example local_time.time_zone = +9 Parameter- Configuration File local_time.time_zone_name .cfg Defines the desired time zone name.
Appendix Parameter- Configuration File local_time.dst_time_type .cfg Configures the DST type. Description Note: It works only if the parameter ―local_time.summer_time‖ is set to 1 (Enabled). Format Integer Default Value Blank Valid values are: Range 0-By Date 1-By Week Example local_time.dst_time_type = 1 Parameter- Configuration File local_time.start_time .cfg Specifies the time to start DST. If ―local_time.
Administrator’s Guide for SIP-T46G IP Phone Range 1to 12/1 to 31/0 to 23 (for By Date) 1 to 12/1 to 5/1 to 7/0 to 23 (for By Week) Example local_time.start_time = 5/20/12 Parameter- Configuration File local_time.end_time .cfg Specifies the time to end DST. If ―local_time.dst_time_type‖ is set to 0 (By Date), use the mapping: MM: 1=Jan, 2=Feb,…, 12=Dec DD:1=the first day in a month,…, 31= the last day in a month HH:0=1am, 1=2am,…, 23=12pm If ―local_time.
Appendix (Enabled). Format Integer Default Value 60 Range -300 to +300 Example local_time.offset_time = 120 Time Format Parameter- Configuration File local_time.time_format .cfg Sets the time format. If set to 0 (12 Hour), the time display uses 12 Description hour format. If set to 1 (24 Hour), the time display uses 24 hour format. Format Integer Default Value 1 Range Example 0-12 Hour 1-24 Hour local_time.
Administrator’s Guide for SIP-T46G IP Phone Example local_time.date_format = 1 Parameter- Configuration File gui_lang.url .cfg Specifies the access URL of the language pack. Note: The language packs you load are Description dependent on available language packs from the provisioning server. You can download the language pack to the phone user interface only.
Appendix Chinese French German Italian Portuguese Spanish Turkish Example lang.wui = French Parameter- Configuration File lang.gui .cfg Description Specifies the language used on the phone user interface. Format Text Default Value English Valid values are: English Chinese French Range German Italian Polish Portuguese Spanish Turkish Example lang.gui = English Parameter- Configuration File features.pound_key.mode .
Administrator’s Guide for SIP-T46G IP Phone Format Integer Default Value 1 Valid values are: Range 0-Disabled 1-# key 2-* key Example features.pound_key.mode = 0 Parameter- Configuration File features.send_key_tone .cfg Enables or disables the IP phone to play a tone when a user presses a send key. If set to 1 (Enabled), the IP phone plays a tone Description when a user presses a send key. Note: It works only if the key tone is enabled. So you should set the parameter ―features.
Appendix Range Not Applicable Example features.hotline_number = 3601 Parameter- Configuration File features.hotline_delay .cfg Specifies the waiting time (in seconds) the IP phone automatically dials out the hotline number. If set to 0 (0s), the IP phone immediately dials out the preconfigured hotline number when Description you lift the handset, press the speakerphone key or press the line key.
Administrator’s Guide for SIP-T46G IP Phone Example features.history_save_display = 0 Parameter- Configuration File features.save_call_history .cfg Enables or disables the IP phone to save call log. Description If set to 0 (Disabled), the IP phone cannot log the dialed calls, received calls, missed calls and the forwarded calls in the call log lists. Format Boolean Default Value 1 Range 0-Disabled 1-Enabled Example features.
Appendix Parameter- Configuration File phone_setting.predial_autodial .cfg Configures live dialpad feature. Description If set to 1 (Enabled), the IP phone automatically dials out the entered phone number without having to press any key. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example phone_setting.predial_autodial = 1 Parameter- Configuration File call_waiting.enable .cfg Enables or disables the call waiting feature.
Administrator’s Guide for SIP-T46G IP Phone audible indicator when receiving a new incoming call during a call. Note: It works only if the parameter ―call_waiting.enable‖ is set to 1 (Enabled). Format Boolean Default Value 1 Range 0-Disabled 1-Enabled Example call_waiting.tone = 1 Parameter- Configuration File auto_redial.enable .cfg Enables or disables the IP phone to automatically redial the called number when it Description is busy.
Appendix Example auto_redial.interval = 30 Parameter- Configuration File auto_redial.times .cfg Sets the redial times for the IP phone. Description The IP phone tries to redial the dialed number as many times as configured till the callee answers the call. Format Integer Default Value 10 Range 1 to 300 Example auto_redial.times = 8 Parameter- Configuration File account.x.auto_answer .cfg Enables or disables the auto answer feature for account X.
Administrator’s Guide for SIP-T46G IP Phone Parameter- Configuration File features.call_completion_enable .cfg Enables or disables the call completion feature. If a user places a call and the callee is temporarily not available to answer the call, Description the call completion feature allows notifying the user when the callee becomes available to receive a call. If set to 1 (Enabled), the caller is notified when the callee becomes available to receive a call.
Appendix Parameter- Configuration File account.x.anonymous_call_onc .cfg ode Sets the anonymous call on code to activate Description the server-side anonymous call feature for account X (optional). X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example account.1.anonymous_call_oncode = *72 Parameter- Configuration File account.x.anonymous_call_offc .
Administrator’s Guide for SIP-T46G IP Phone 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_o .cfg ncode Sets the anonymous call rejection on code to Description activate the server-side anonymous call rejection feature for account X (optional). X ranges from 1 to 6. Format String Default Value Blank Range Not Applicable Example account.1.
Appendix Return Message When DND Parameter- Configuration File features.dnd_refuse_code .cfg Defines 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 the DND feature.
Administrator’s Guide for SIP-T46G IP Phone DND in Phone Mode Parameter- Configuration File features.dnd.enable .cfg Enables or disables the 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 .cfg Description Sets the DND on code to activate the server-side DND feature.
Appendix account X. 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 Sets 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-T46G IP Phone Parameter- Configuration File features.busy_tone_delay .cfg Configures a period of time (in seconds) for which the busy tone is audible on the IP phone. When one party releases the call, a busy tone Description is audible to the other party indicating that the call connection breaks. If set to 3 (3s), a busy tone is audible for 3 seconds on the IP phone.
Appendix Example features.normal_refuse_code = 480 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 0 Range 0-Disabled 1-Enabled Example phone_setting.is_deal180 = 1 Parameter- Configuration File sip.
Administrator’s Guide for SIP-T46G IP Phone 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 Not Applicable Example account.1.advanced.timer_t1 = 1 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 Not Applicable Example account.1.advanced.timer_t4 = 10 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-T46G IP Phone Example account.1.session_timer.expires = 300 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.session_timer.
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 = 60 Parameter- Configuration File sip.rfc2543_hold .cfg Specifies whether RFC 2543 (c=0.0.0.0) outgoing hold signaling is used.
Administrator’s Guide for SIP-T46G IP Phone 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 < y000000000028 >.cfg Enables or disables the always forward feature. Description If set to 1 (Enabled), incoming call are forwarded to the destination number immediately. Format Boolean Default Value 0 Range 1-Enabled Example forward.always.
Appendix Default Value Blank Range Not Applicable Example forward.always.on_code = *72 Parameter- Configuration File forward.always.off_code < y000000000028 >.cfg Sets 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 < y000000000028 >.cfg Enables or disables the busy forward feature.
Administrator’s Guide for SIP-T46G IP Phone Example forward.busy.target = 3602 Parameter- Configuration File forward.busy.on_code < y000000000028 >.cfg Description Sets the busy forward on code to 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 < y000000000028 >.
Appendix Parameter- Configuration File forward.no_answer.target < y000000000028 >.cfg Description Defines 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 < y000000000028 >.cfg Defines a period of ring time to wait before Description forwarding the incoming call.
Administrator’s Guide for SIP-T46G IP Phone 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 .cfg Enables or disables the always forward feature for account X. Description If set to 1 (Enabled), incoming calls to the account X are forwarded to the destination number immediately. X ranges from 1 to 6.
Appendix Parameter- Configuration File account.x.always_fwd.on_code .cfg Sets the always forward on code activate Description 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 .cfg Sets the always forward off code to Description deactivate the server-side always forward feature for account X.
Administrator’s Guide for SIP-T46G IP Phone Example account.1.busy_fwd.enable = 1 Parameter- Configuration File account.x.busy_fwd.target .cfg Defines the destination number of the busy Description 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 the 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-T46G IP Phone Example account.1.timeout_fwd.timeout = 5 Parameter- Configuration File account.x.timeout_fwd.on_code .cfg Sets the no answer forward on code to Description 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 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-T46G IP Phone Parameter- Configuration File account.x.conf_type .cfg Defines the conference type for account X. If set to 0 (Local), conferences are set up on Description the IP phone locally. If set to 2 (Network Conference), conferences are set up by the server. X ranges from 1 to 6. Format Integer Default Value 0 Valid values are: Range 0-Local 2-Network Conference Example account.1.conf_type = 2 Parameter- Configuration File account.x.conf_uri .
Appendix Parameter- Configuration File transfer.tran_others_after_conf_e .cfg nable Enables or disables the 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.tran_others_after_conf_enable = 1 Phone Basis Parameter- Configuration File features.
Administrator’s Guide for SIP-T46G IP Phone Configures the directed call pickup code on a phone basis. Description Note: The directed call pickup code configured on a per-account 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-account Basis Parameter- Configuration File account.x.direct_pickup_code .
Appendix Default Value Range 0 0-Disabled 1-Enabled Example features.pickup.group_pickup_enable = 1 Parameter- Configuration File features.pickup.group_pickup_co .cfg de Configures the group call pickup code on a phone basis. Description Note: The group call pickup code configured on a per-account basis takes precedence over that configured on a phone basis. Format String Default Value Blank Range Not Applicable Example features.pickup.
Administrator’s Guide for SIP-T46G IP Phone Parameter- Configuration File account.x.dialoginfo_callpickup .cfg Configures the Dialog-Info Call Pickup feature for account X. Description If set to 1 (Enabled), call pickup is implemented through SIP signals. X ranges from 1 to 6. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example account.1.dialoginfo_callpickup = 1 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 = 90 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-T46G IP Phone 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 = 2 Parameter- Configuration File account.x.
Administrator’s Guide for SIP-T46G IP Phone 2-SIP INFO 3-AUTO+SIP INFO Example account.1.dtmf.type = 2 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 Example features.dtmf.repetition = 2 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-T46G IP Phone the transfer to screen when pressing the transfer key during a call. If set to 1 (Enabled), the IP phone transmits the specified DTMF digits to the server when pressing the transfer key during a call, and then complete the transfer. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example features.dtmf.replace_tran = 1 Parameter- Configuration File features.dtmf.transfer .
Appendix Format Boolean Default Value 1 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.
Administrator’s Guide for SIP-T46G IP Phone Example features.intercom.tone = 1 Parameter- Configuration File features.intercom.barge .cfg Enables or disables the IP phone to automatically answer an incoming intercom call while there is already an active call on the IP phone. If set to 0 (Disabled), the IP phone handles an incoming intercom call like a waiting call Description while there is already an active call on the IP phone.
Appendix Parameter- Configuration File account.x.alert_info_url_enable .cfg Enables or disables the distinctive ring Description tones feature for account X. X ranges from 1 to 6. Format Boolean Default Value 0 Range Example 0-Enabled 1-Disabled account.1.alert_info_url_enable = 1 Parameter- Configuration File distinctive_ring_tones.alert_info.x.tex .cfg t Specifies the texts to map the keywords Description contained in the SIP header. X ranges from 1 to 10.
Administrator’s Guide for SIP-T46G IP Phone 1-Ring1.wav 2-Ring2.wav 3-Ring3.wav 4-Ring4.wav 5-Ring5.wav 6-Ring6.wav 7-Ring7.wav 8-Ring8.wav distinctive_ring_tones.alert_info.1.ringer Example =2 Parameter- Configuration File voice.tone.country .cfg Description Configures the tone type for the IP phone.
Appendix Portugal Spain Switzerland Sweden Russia United States Chile Czech ETSI Example voice.tone.country = Austria Parameter- Configuration File voice.tone.dial .cfg voice.tone.ring voice.tone.busy voice.tone.congestion voice.tone.callwaiting voice.tone.dialrecall voice.tone.record voice.tone.info voice.tone.stutter voice.tone.message voice.tone.autoanswer Customizes the tone for each condition.
Administrator’s Guide for SIP-T46G IP Phone Note: It works only if the parameter ―voice.tone.country‖ is set to Custom. Format F/D or !F/D Default Value Blank Range Not Applicable Example voice.tone.dial = 800+200/1000, 0/100, 500/1200, 500+600+950+1500/5000 Parameter- Configuration File features.remote_phonebook.ena .cfg ble Enables or disables the IP phone to perform a Description remote phonebook search when receiving an incoming call.
Appendix Parameter- Configuration File ldap.enable .cfg Description Enables or disables the LDAP feature on the IP phone. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example ldap.enable =1 Parameter- Configuration File ldap.name_filter .cfg Specifies the name attribute for LDAP searching. The Description ―*‖ symbol in the filter stands for any character.
Administrator’s Guide for SIP-T46G IP Phone (|(telephoneNumber=%)(Mobile=%)(ipPhone=%)) 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 phone LCD screen. Parameter- Configuration File ldap.host .cfg Description Specifies the domain name or IP address of the LDAP server. Format IP Address or Domain Name Default Value 0.0.0.0 Range Not Applicable Example ldap.host = 192.168.1.
Appendix Parameter- Configuration File ldap.user .cfg Specifies the user name uses to login the LDAP server. Description This parameter can be left blank in case the server allows anonymous to login. Otherwise you will need to provide the username to access the LDAP server. Format String Default Value Blank Range Not Applicable Example ldap.user = cn=manager,dc=yealink,dc=cn Parameter- Configuration File ldap.password .
Administrator’s Guide for SIP-T46G IP Phone Example ldap.max_hits = 60 Parameter- Configuration File ldap.name_attr .cfg Specifies the name attributes of each record to be Description returned by the LDAP server. It compresses the search results. You can configure multiple name attributes separated by space. Format String Default Value Blank Range Not Applicable Example ldap.name_attr = cn sn Parameter- Configuration File ldap.numb_attr .
Appendix LCD screen. Parameter- Configuration File ldap.version .cfg Specifies the LDAP protocol version supported by the Description IP phone. Make sure the protocol value corresponds with the version assigned on the LDAP server. Format Integer Default Value 3 Range 2 or 3 Example ldap.version = 3 Parameter- Configuration File ldap.call_in_lookup .cfg Description Enables or disables the IP phone to perform an LDAP search when receiving an incoming call.
Administrator’s Guide for SIP-T46G IP Phone Visual and Audio Alert for BLF Pickup Parameter- Configuration File features.pickup.blf_visua .cfg l_enable Enables or disables the IP phone to display a visual Description prompt when the monitored user receives an incoming call. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example features.pickup.blf_visual_enable = 1 Parameter- Configuration File features.pickup.blf_audi .
Appendix Parameter- Configuration File account.x.music_server_ .cfg uri Specifies the Music on Hold server address. Examples for valid values: <10.1.3.165>, 10.1.3.165, sip:moh@ucap.com, , Description or yealink.com. X ranges from 1 to 6. Note: The DNS query in this parameter only supports A query. Format String Default Value Blank Range Not Applicable Example account.1.music_server_uri =<10.1.3.165> Parameter- Configuration File account.X.acd.
Administrator’s Guide for SIP-T46G IP Phone Default Value Value 0 0- Disabled 1- Enabled Example account.X.acd.available = 1 Parameter- Configuration File account.X.acd.user_id .cfg Description Configures the user ID used to log in the ACD system. X ranges from 1 to 6. Format String Default Value Blank Value Not Applicable Example account.X.acd.user_id = 3606 Parameter- Configuration File account.X.acd.password .
Appendix Parameter- Configuration File acd.auto_available_timer .cfg Description Specifies the length of time (in seconds) before the IP phone state is automatically reset to ―available‖. Format Integer Default Value 60 Value 0 to 120 Example acd.auto_available_timer = 80 Parameter- Configuration File account.x.subscribe_mwi .cfg Enables or disables the IP phone to subscribe the message waiting indicator for account X.
Administrator’s Guide for SIP-T46G IP Phone (Enabled). Format Integer Default Value 3600 Value 0 to 84600 Example account.1.subscribe_mwi_expires = 3600 Parameter- Configuration File account.X.subscribe_mwi_to_vm .cfg Enables or disables a subscription to the Description voice mail number for MWI service for account X. X ranges from 1 to 6. Format Boolean Default Value 0 Value 0-Disabled 1-Enabled Example account.1.
Appendix Parameter- Configuration File multicast.codec .cfg Description Specifies a multicast codec for the IP phone to use to send an RTP stream. Format string Default Value G722 Valid values are: Range PCMU PCMA G729 G722 G726-16 G726-24 G726-32 G726-40 G723_53 Example multicast.codec = G722 Parameter- Configuration File multicast.receive_priority.enable .
Administrator’s Guide for SIP-T46G IP Phone Example multicast.receive_priority.enable =1 Parameter- Configuration File multicast.receive_priority.priority < y000000000028 >.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. Format Integer Default Value 10 Range 0 to10 Example multicast.receive_priority.
Appendix Default Value Blank Range Not Applicable Example multicast.listen_address.1.ip_address = 224.5.6.20:10008 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.outgoing_call = action_url.call_established = action_url.dnd_on = action_url.dnd_off = action_url.always_fwd_on = action_url.always_fwd_off = action_url.
Administrator’s Guide for SIP-T46G IP Phone action_url.ip_change = action_url.forward_incoming_call = action_url.reject_incoming_call = action_url.call_interrupt = action_url.call_remote_busy = action_url.call_remote_canceled = action_url.answer_new_incoming_ call = action_url.reject_new_incoming_ca ll= action_url.cancel_callout = action_url.remote_busy = action_url.transfer_finished = action_url.transfer_failed = Specifies the URL for the predefined event.
Appendix Parameter- Configuration File features.action_uri_limit_ip .cfg Specifies the address(es) from which Action URI will be accepted. For discontinuous IP addresses, each IP address is separated by comma. For continuous IP addresses, the format likes *.*.*.* and the ―*‖ stands for the values 0~255. Description For example: 10.10.*.* stands for the IP addresses that range from 10.10.0.0 to 10.10.255.255. If left blank, the IP phone cannot receive or handle any HTTP GET request.
Administrator’s Guide for SIP-T46G IP Phone Example account.1.naptr_build = 3 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 = 1 Parameter- Configuration File account.x.fallback.timeout .
Appendix for the service type and port. X ranges from 1 to 6. Format Integer 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.sip_server.y.address .cfg Configures the IP address or domain name Description of the SIP server. X ranges from 1 to 6. Y ranges from 1 to 2. Format IP Address or Domain Name Default Value Blank Range Not Applicable Example account.1.sip_server.1.address = as.
Administrator’s Guide for SIP-T46G IP Phone Parameter- Configuration File account.1.sip_server.1.expires .cfg Configures the registration expires (in Description seconds). X ranges from 1 to 6. Y ranges from 1 to 2. Format Integer Default Value 3600 Range 30 to 2147483647 Example account.1.sip_server.1.expires = 3500 Parameter- Configuration File account.x.sip_server.y.transport_ty .cfg pe Configures the transport type for the SIP Description server. X ranges from 1 to 6.
Appendix Default Value 3 Range 0 to 65535 Example account.1.sip_server.1.retry_counts = 3 Parameter- Configuration File account.x.sip_server.y.failback_mo .cfg de Configures the way in which the phone fails back to the primary server for call control Description when in the failover mode. X ranges from 1 to 6. Y ranges from 1 to 2.
Administrator’s Guide for SIP-T46G IP Phone X ranges from 1 to 6. Y ranges from 1 to 2. Format Integer Default Value 3600 Range 0 to 65535 Example account.1.sip_server.1.failback_timeout = 3200 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.
Appendix Example network.lldp.enable = 1 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-T46G IP Phone Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 0 Range 0 to 4094 Example network.vlan.internet_port_vid = 1 Parameter- Configuration File network.vlan.internet_port_priority .cfg Specifies the priority value used for passing VLAN packets. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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 Specifies the priority value used for passing VLAN packets.
Administrator’s Guide for SIP-T46G IP Phone Example network.vlan.dhcp_enable = 1 Parameter- Configuration File network.vlan.dhcp_option .cfg Description Specifies the option of the OpenVPN tar package. Format String Default Value Blank Range Not Applicable Example network.vlan.dhcp_option = 132,140, Parameter- Configuration File network.vpn_enable .cfg Enables or disables the VPN feature on the IP phone.
Appendix 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 = 50 Parameter- Configuration File network.qos.signaltos .cfg Configures the DSCP for SIP packets.
Administrator’s Guide for SIP-T46G IP Phone Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example account.1.nat.nat_traversal = 1 Parameter- Configuration File account.x.nat.stun_server .cfg Specifies 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.nat.
Appendix Format Boolean Default Value 1 Range 0-Disabled 1-Enabled Example network.snmp.enable = 0 Parameter- Configuration File network.snmp.port .cfg Specifies 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 161 Range 0 to 65535 Example network.snmp.port = 1008 Parameter- Configuration File network.snmp.trust_ip .
Administrator’s Guide for SIP-T46G IP Phone Example network.snmp.trust_ip = 192.168.1.50 server@manager.com Parameter- Configuration File network.802_1x.mode .cfg Specifies 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.
Appendix Parameter- Configuration File network.802_1x.md5_password .cfg Enters the password used for authenticating the IP phone. Note: If you change this parameter, the IP Description phone will reboot to make the change take effect. 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.
Administrator’s Guide for SIP-T46G IP Phone Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is only applicable to the EAP-TLS protocol. The format of the certificate must be *.pem or *.cer. Format String Default Value Blank Range Not Applicable network.802_1x.client_cert_url = Example http://192.168.1.10/ client.pem Parameter- Configuration File managementserver.enable .cfg Enables or disables the TR-069 feature on the IP phone.
Appendix Example managementserver.username = user1 Parameter- Configuration File managementserver.password .cfg Enters the password to authenticate with the ACS. This string is set to the empty string if no Description 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.
Administrator’s Guide for SIP-T46G IP Phone Range Example Not Applicable managementserver.connection_request_usern ame = acsuser Parameter- Configuration File managementserver.connection .cfg _request_password Sets 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 form_interval Sets the interval (in seconds) to report its configuration information to the ACS. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Integer Default Value 60 Range Not Applicable Example managementserver.periodic_inform_interval = 120 Parameter- Configuration File network.ip_address_mode .cfg Specifies the IP address mode.
Administrator’s Guide for SIP-T46G IP Phone Default Value 0 Valid values are: Range 0-DHCP 1-Static Example network.ipv6_internet_port.type = 1 Parameter- Configuration File network.ipv6_internet_port.ip .cfg Configures the IPv6 address. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank Range Not Applicable Example network.ipv6_internet_port.
Appendix 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 .cfg Configures the primary DNS server when the Internet port type is defined as Static IP Description Address. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Administrator’s Guide for SIP-T46G IP Phone Parameter- Configuration File network.ipv6_icmp_v6.enable .cfg Enables or disables the ICMPv6 feature. If set to 1 (enabled), the IP phone obtains the parameters of the IPv6 from the ICMPv6 Description protocol. Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Boolean Default Value 1 Range Example 0-Disabled 1-Enabled network.ipv6_icmp_v6.
Appendix Parameter- Configuration File features.headset_training .cfg Enables or disables the 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-T46G IP Phone When Y=9, the default value is 0; When Y=10, the default value is 0; When Y=11, the default value is 0; When Y=12, the default value is 0; When Y=13, 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 Specifies the codec for account X to use. Description X ranges from 1 to 6. Y ranges from 0 to 13.
Appendix Example G726_24 G726_32 G726_40 iLBC iLBC_13_3 iLBC_15_2 GSM account.1.codec.1.payload_type = G723_53 Parameter- Configuration File account.X.codec.Y.priority .cfg Specifies the priority for the codec. Description X ranges from 1 to 6. Y ranges from 0 to 13.
Administrator’s Guide for SIP-T46G IP Phone X ranges from 1 to 6. Y ranges from 0 to 13.
Appendix Parameter- Configuration File voice.echo_cancellation .cfg Description Enables or disables the AEC feature on the IP phone. Format Boolean Default Value 1 Range 0-Disabled 1-Enabled Example voice.echo_cancellation = 1 Parameter- Configuration File voice.vad .cfg Description Enables or disables the VAD feature on the IP phone. Format Boolean Default Value 0 Range 0-Disabled 1-Enabled Example voice.
Administrator’s Guide for SIP-T46G IP Phone Parameter- Configuration File voice.jib.adaptive .cfg Description Configures the type of jitter buffer. Format Integer Default Value 1 Valid values are: Range 0-Fixed 1-Adaptive Example voice.jib.adaptive = 1 Parameter- Configuration File voice.jib.min .cfg Configures the minimum delay time for jitter Description buffer. Note: It works only if the parameter ―voice.jib.adaptive‖ is set to 1 (Adaptive).
Appendix Parameter- Configuration File voice.jib.normal .cfg Configures the fixed delay time for jitter Description buffer. Note: It works only if the parameter ―voice.jib.adaptive‖ is set to 0 (Fixed). Format Integer Default Value 120 Range Not Applicable Example voice.jib.mormal = 100 Parameter- Configuration File account.x.transport .cfg Configures the transport type for account X.
Administrator’s Guide for SIP-T46G IP Phone 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 security.trust_certificates = 1 Parameter- Configuration File security.ca_cert .cfg Specifies the type of certificates the IP phone used to authenticate the connecting Description server. Note: If you change this parameter, the IP phone will reboot to make the change take effect.
Appendix Example security.cn_validation = 1 Parameter- Configuration File security.dev_cert .cfg Specifies the type of certificates the IP phone sends for authentication. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format Boolean Default Value 0 Range 0-Default certificates 1-Custom certificates Example security.dev_cert = 1 Parameter- Configuration File trusted_certificates.url .
Administrator’s Guide for SIP-T46G IP Phone Default Value Blank Range Not Applicable Example server_certificates.url = http://192.168.1.20/ca.pem Parameter- Configuration File account.x.srtp_encryption .cfg Configures whether to use voice encryption service. If the set to 1 (Forced), the IP phone is Description forced to using SRTP during a call. If set to 2 (Negotiated), the IP phone will negotiate with the other IP phone what type of encryption to utilize for the session.
Appendix Example auto_provision.aes_key_16.com = 0123456789abcdef Parameter- Configuration File auto_provision.aes_key_16.mac .cfg Description Format Configures the AES key which is used to encrypt or decrypt the .cfg file. String () ><| "& cannot be included. Default Value Blank Range 16 characters Example auto_provision.aes_key_16.mac = 0123456789abmins Parameter- Configuration File auto_provision.mode .
Administrator’s Guide for SIP-T46G IP Phone ―auto_provision.mode‖ is set to 4(Repeatedly) or 6 (Power on + Repeatedly). Format Integer Default Value 1440 Range 1 to 43200 Example auto_provision.schedule.periodic_minute = 1000 Parameter- Configuration File auto_provision.schedule.time_from < y000000000028 >.cfg Configures the start 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.
Appendix Parameter- Configuration File auto_provision.schedule.dayofwe < y000000000028>.cfg ek Description Defines the desired day(s) of a week for the IP phone to check new configuration. Format Integer Default Value 0123456 Valid values are: 0-Sunday 1-Monday Range 2-Tuesday 3-Wednesday 4-Thursday 5-Friday 6-Saturday Example auto_provision.schedule.time_to = 123 Parameter- Configuration File firmware.url .cfg Description Specifies the access URL of the firmware.
Administrator’s Guide for SIP-T46G IP Phone Default Value Blank Range Not Applicable Example dialplan_replace_rule.url = http://192.168.10.25/dialplan.xml Parameter- Configuration File dialplan_dialnow.url .cfg Description Specifies 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 Parameter- Configuration File custom_softkey_call_failed.url .
Appendix Parameter- Configuration File custom_softkey_call_in.url .cfg Specifies the access URL of the customized Description file for the soft key presented on the phone LCD screen when in the CallIn state. 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.
Administrator’s Guide for SIP-T46G IP Phone Format URL Default Value Not Applicable Range Not Applicable 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 .
Appendix ―XMLfiles‖ directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_talking.url = http://10.2.8.16:8080/XMLfiles/Talking.xml Parameter- Configuration File local_contact.data.url .cfg Description Specifies 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 Parameter- Configuration File remote_phonebook.data.x.url .
Administrator’s Guide for SIP-T46G IP Phone image. Format URL Default Value Blank Range Not Applicable Example wallpaper_upload.url = http://192.168.10.25/wallpaper.jpg Parameter- Configuration File syslog.server .cfg Specifies the IP address of the syslog server where to export the log files. Description Note: If you change this parameter, the IP phone will reboot to make the change take effect. Format IP Address Default Value Blank Range Not Applicable Example syslog.
Appendix Parameter- Configuration File watch_dog.enable .cfg Description Enables or disables the Watch Dog feature. Format Boolean Default Value 1 Range Example 0-Disabled 1-Enabled watch_dog.enable = 1 This section provides the DSS key parameters you can configure on the IP phone. Various key features can be assigned to the DSS key. The parameters of the DSS key are detailed in the following: Parameter- Configuration File linekey.x.line .
Administrator’s Guide for SIP-T46G IP Phone LDAP Conference Forward Hold DND Call Return SMS Record URL Record Multicast Paging Group Listening Private Hold Zero Touch URL Keypad Lock Favorite Format Integer Default Value 0 (Auto) Range Valid values are: 0 to 6 Example linekey.1.line = 2 Parameter- Configuration File linekey.x.value .cfg Description Specifies the value for some key features. X ranges from 1 to 27.
Appendix feature. This parameter only applies to the BLF feature. X ranges from 1 to 27. Format String Default Value Blank Range Not Applicable Example linekey.1.pickup_value = *88 Parameter- Configuration File linekey.x.type .cfg Specifies the key feature for the line key. X ranges from 1 to 27.
Administrator’s Guide for SIP-T46G IP Phone Prefix Zero Touch ACD Local Group Keypad Lock Custom Button Favorite Format Integer Default Value 0 (N/A) Valid values are: 0-N/A(default for line key 7-27) 1-Conference 2-Forward 3-Transfer 4-Hold 5-DND 7-Call Return 8-SMS 9-Call Pickup 10-Call Park 11-DTMF 12-Voicemail 13-SpeedDial 14-Intercom 15-Line(default for line key 1-6) Range 16-BLF 17-URL 18-Group Listening 22-XML Group 23-Group Pickup 24-Multicast Paging 25-Record 27-XM
Appendix Example linekey.1.type = 8 Parameter- Configuration File linekey.x.xml_phonebook .cfg Specifies the desired phonebook when multiple phonebooks are configured on the Description IP phone. This parameter only applies to the Local Group/XML Group features. X ranges from 1 to 27. Format Integer Default Value 0 Range Not Applicable Specify the second phonebook when Example there are three BroadSoft groups are configured on the IP phone. linekey.1.
Administrator’s Guide for SIP-T46G IP Phone IP phone. The digit 5 stands for the key type DND. X ranges from 1 to 27. Format Integer Value 5 Example linekey.1.type = 5 Directed Call Pickup Key Parameter- Configuration File linekey.x.type .cfg Configures a line key to be directed call pickup key on the IP phone. Description The digit 9 stands for the key type Call Pickup. X ranges from 1 to 27. Format Integer Value 9 Example linekey.1.
Appendix X ranges from 1 to 27. Format String Range Not Applicable Example linekey.1.value = *971001 Group Call Pickup Key Parameter- Configuration File linekey.x.type .cfg Configures a line key to be group call pickup key on the IP phone. Description The digit 23 stands for the key type Group Pickup. X ranges from 1 to 10. Format Integer Value 23 Example linekey.1.type = 23 Parameter- Configuration File linekey.x.line .
Administrator’s Guide for SIP-T46G IP Phone Range Not Applicable Example linekey.1.value = *98 Call Return Key Parameter- Configuration File linekey.x.type .cfg Configures a line key to be call return key on the IP phone. Description The digit 7 stands for the key type Call Return. X ranges from 1 to 27. Format Integer Value 7 Example linekey.2.type = 7 Call Park Key Parameter- Configuration File linekey.x.type .
Appendix Range Valid values are: 0 to 5 Example linekey.2.line = 0 Parameter- Configuration File linekey.x.value .cfg Description Specifies the call park feature code. X ranges from 1 to 27. Format String Range Not Applicable Example linekey.2.value = *99 Intercom Key Parameter- Configuration File linekey.x.type .cfg Configures a line key to be the intercom key. Description The digit 14 stands for the key type Intercom. X ranges from 1 to 27.
Administrator’s Guide for SIP-T46G IP Phone Parameter- Configuration File linekey.x.value .cfg Description Specifies the intercom number. X ranges from 1 to 27. Format String Range Not Applicable Example linekey.2.value = 1008 LDAP Key Parameter- Configuration File linekey.x.type .cfg Configures a line key to be LDAP key on the Description IP phone. The digit 38 stands for the key type LDAP. X ranges from 1 to 27. Format Integer Value 38 Example linekey.
Appendix key. X ranges from 1 to 27. Format Range Integer Valid values are: 0 to 5 Example linekey.3.line = 2 Parameter- Configuration File linekey.x.value .cfg Description Specifies the number of the monitored user. X ranges from 1 to 27. Format String Range Not Applicable Example linekey.3.value = 1008 Parameter- Configuration File linekey.x.pickup_value .cfg Specifies the pickup code for the BLF feature.
Administrator’s Guide for SIP-T46G IP Phone Format Integer Value 42 Example linekey.2.type = 42 Parameter- Configuration File linekey.x.line .cfg Specifies the desired line to apply the ACD Description key. X ranges from 1 to 27. Format Range Example Integer Valid values are: 0 to 5 linekey.2.line = 1 Multicast Paging Key Parameter- Configuration File linekey.x.type .cfg Configures a line key to be a multicast paging key on the IP phone.
Appendix Example linekey.3.value = 224.5.5.6:10008 Record Key Parameter- Configuration File linekey.x.type .cfg Configures a line key to be a record key on Description the IP phone. The digit 25 stands for the key type Record. X ranges from 1 to 27. Format Integer Value 25 Example linekey.2.type = 25 URL Record Key Parameter- Configuration File linekey.x.type .cfg Configures a line key to be a URL record key on the IP phone.
Administrator’s Guide for SIP-T46G IP Phone http://10.1.2.224/phonerecording.cgi Hot Desking Key Parameter- Configuration File linekey.x.type .cfg Configures a line key to be a hot desking key on the IP phone. Description The digit 34 stands for the key type hot desking. X ranges from 1 to 27. Format Integer Value 34 Example linekey.2.type = 34 This section describes how the Yealink SIP-T46G IP phones comply with the IETF definition of SIP as described in RFC 3261.
Appendix RFC 3264—An Offer/Answer Model with the Session Description Protocol (SDP) RFC 3265—Session Initiation Protocol (SIP) - Specific Event Notification RFC 3311—The Session Initiation Protocol (SIP) UPDATE Method RFC 3325—SIP Asserted Identity RFC 3515—The Session Initiation Protocol (SIP) Refer Method RFC 3555—MIME Type of RTP Payload Formats RFC 3611—RTP Control Protocol Extended reports (RTCP XR) RFC 3665—Session Initiation Protocol (SIP) Basic Call Flow Examples
Administrator’s Guide for SIP-T46G IP Phone To find the applicable Request for Comments (RFC) document, go to http://www.ietf.org/rfc.html and enter the RFC number. The following SIP request messages are supported: Method REGISTER Supported Notes Yes The Yealink SIP-T46G IP phones support mid-call INVITE Yes changes such as putting a call on hold as signaled by a new INVITE that contains an existing Call-ID.
Appendix Method Supported 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-Authenticate Yes Proxy-Authorization Yes RAck Yes Record-Route Yes Refer-To Yes Referred-By Yes Remote-Party-ID Yes Replaces Yes Require Yes Route Yes RSeq Yes Session-Expires Yes S
Administrator’s Guide for SIP-T46G IP Phone Method Supported To Yes User-Agent Yes Via Yes Notes The following SIP responses are supported: 1xx Response—Information Responses 1xx Response Supported 100 Trying Yes 180 Ringing Yes 181 Call Is Being Forwarded Yes 183 Session Progress Yes Notes 2xx Response—Successful Responses 2xx Response Supported 200 OK Yes 202 Accepted Yes Notes In REFER transfer.
Appendix 4xx Response Supported 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 411 Length Required No 413 Request Entity Too Large No 414 Request-URI Too Long Yes 415 Unsupported Media Type Yes 416 Unsupported URI Scheme No 420 Bad Extension No 421 Extension Required No 423 Interval Too Brief Yes 480 Temporarily Unavaila
Administrator’s Guide for SIP-T46G IP Phone 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—Global Responses 6xx Response 600 Busy Everywhere Yes 603 Decline Yes 604 Does Not Exist Anywhere No 606 Not Acceptable No SDP Headers v—Protocol version o—Owner/creator and session identifier Notes Supported Y
Appendix SIP uses six request methods: INVITE—Indicates a user is being invited to participate in a call session. ACK—Confirms that the client has received a final response to an INVITE request. BYE—Terminates a call and can be sent by either the caller or the callee. CANCEL—Cancels any pending searches but does not terminate a call that has already been accepted. OPTIONS—Queries the capabilities of servers.
Administrator’s Guide for SIP-T46G IP Phone The following figure illustrates the scenario of a successful call. In this scenario, the two end users are User A and User B. User A and User B are located at the Yealink SIP IP phones. The call flow scenario is as follows: 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.
Appendix 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.
Administrator’s Guide for SIP-T46G IP Phone 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 message to User A. The 200 OK A 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.
Appendix 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.
Administrator’s Guide for SIP-T46G IP Phone 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 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. The proxy server forwards the SIP ACK F8 ACK—Proxy Server to User B to User B to indicate that the 486 Busy Here message has already been received.
Administrator’s Guide for SIP-T46G IP Phone 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 the 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. INVITE B F2.
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-T46G IP Phone Step F7 Action Description User B User A wants to disconnect the call. 200 OK—User B to Proxy User B sends a SIP 200 OK response to Server the proxy server. The SIP 200 OK response indicates that User B has received the CANCEL request. F8 200 OK—Proxy Server to User The proxy server forwards the SIP 200 A OK response to notify User A that the CANCEL request has been processed successfully.
Appendix The following figure illustrates a successful call setup and call hold. In this scenario, the two end users are User A and User B. User A and User B are located at the Yealink SIP IP phones. The call flow scenario is as follows: 1. User A calls User B. 2. User B answers the call. 3. User A puts User B on hold. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9. INVITE B (sendonly) F10.
Administrator’s Guide for SIP-T46G IP Phone Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Appendix Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server. The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
Administrator’s Guide for SIP-T46G IP Phone 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. INVITE B ( sendonly ) F14. INVITE B ( sendonly ) F15.
Appendix Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Administrator’s Guide for SIP-T46G IP Phone Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server, The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active. User C sends a SIP INVITE message to the proxy server.
Appendix Step Action Description User A sends a mid-call INVITE request F13 INVITE—User A to Proxy to the proxy server with new SDP Server session parameters, which are used to place the call on hold. F14 INVITE—Proxy Server to User The proxy server forwards the mid-call B INVITE message to User B. User B sends a 200 OK to the proxy F15 200 OK—User B to Proxy server. The 200 OK response indicates Server that the INVITE was successfully processed.
Administrator’s Guide for SIP-T46G IP Phone 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.
Appendix User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9. REFER F10. 202 Accepted F11. REFER F12. 202 Accepted F17. BYE F18. BYE F19. 200 OK F20. 200 OK F21. INVITE C F22. INVITE C F23. 180 Ringing F24. 180 Ringing F25. 200 OK F26. 200 OK F27. ACK F28.
Administrator’s Guide for SIP-T46G IP Phone Step Action Description User A sends an INVITE message to the proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Appendix Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server, The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active. User B sends a REFER message to the F9 REFER—User B to Proxy Server proxy server.
Administrator’s Guide for SIP-T46G IP Phone Step Action Description requests the call. F18 INVITE—Proxy Server to User The proxy server maps the SIP URI in the C To field to User C. User C sends a SIP 180 Ringing F19 180 Ringing—User C to Proxy response to the proxy server. The 180 Server Ringing response indicates that the user is being alerted. The proxy server forwards the 180 F20 180 Ringing—Proxy Server to Ringing response to User A.
Appendix 5. User A transfers the call to User C. Call is established between User B and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9. INVITE B （sendonly） F10. INVITE B （sendonly） F11. 200 OK F12. 200 OK F13. ACK F14. ACK F15. INVITE C F16. INVITE C F17. 180 Ringing F18. 180 Ringing F19. 200 OK F20. 200 OK F21. ACK F22. ACK 2-way RTP channel established F23. REFER F24.
Administrator’s Guide for SIP-T46G IP Phone Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Appendix Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server, The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
Administrator’s Guide for SIP-T46G IP Phone Step Action C Description sends the INVITE request to User C. User C sends a SIP 180 Ringing F17 180 Ringing—User C to Proxy response to the proxy server. The 180 Server Ringing response indicates that the user is being alerted. The proxy server forwards the 180 F18 180 Ringing—Proxy Server to Ringing response to User A. User A User A hears the ring-back tone indicating that User C is being alerted.
Appendix Step Action Description response indicates that User B accepts the transfer. User A terminates the call session by F27 BYE—User A to Proxy Server sending a SIP BYE request to the proxy server. The BYE request indicates that User A wants to release the call. F28 BYE—Proxy Server to User B The proxy server forwards the BYE request to User B. User B sends a SIP 200 OK response to F29 200OK—User B to Proxy the proxy server.
Administrator’s Guide for SIP-T46G IP Phone 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.
Appendix Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of the User B is inserted in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Administrator’s Guide for SIP-T46G IP Phone Step Action Description User A sends a SIP INVITE request to the F7 INVITE—User A to Proxy Server proxy server. In the INVITE request, a unique Call-ID is generated and the Contact-URI field indicates that User A requested the call. F8 INVITE—Proxy Server to User C The proxy server maps the SIP URI in the To field to User C. The proxy server sends the SIP INVITE request to User C.
Appendix The following figure illustrates successful call forwarding between Yealink SIP IP phones in which User B has enabled busy call forward. The incoming call is forwarded to User C when User B is busy. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: 1. User B enables busy call forward, and the destination number is User C. 2. User A calls User B. 3.
Administrator’s Guide for SIP-T46G IP Phone Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Appendix Step Action Description ACK message. F7 302 Move Temporarily—Proxy The proxy server forwards the 302 Server to User A Moved Temporarily message to User A. User A sends a SIP ACK to the proxy F8 ACK—User A to Proxy Server server. The ACK message notifies the proxy server that User A has received the ACK message. User A sends a SIP INVITE request to the F9 INVITE—User A to Proxy Server proxy server.
Administrator’s Guide for SIP-T46G IP Phone The following figure illustrates successful call forwarding between Yealink SIP IP phones in which User B has enabled no answer call forward. The incoming call is forwarded to User C when User B does not answer the incoming call after a period of time. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: 1.
Appendix Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: The IP address of User B is inserted in the Request-URI field. User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
Administrator’s Guide for SIP-T46G IP Phone Step Action Description ACK message. F7 302 Move Temporarily—Proxy The proxy server forwards the 302 Server to User A Moved Temporarily message to User A. User A sends a SIP ACK to the proxy F8 ACK—User A to Proxy Server server. The ACK message notifies the proxy server that User A has received the ACK message. User A sends a SIP INVITE request to the F9 INVITE—User A to Proxy Server proxy server.
Appendix The following figure illustrates successful 3-way calling between Yealink SIP-T46G IP phones in which User A mixes two RTP channels and therefore establishes a conference between User B and User C. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: 1. User A calls User B. 2. User B answers the call. 3. User A puts User B on hold. 4.
Administrator’s Guide for SIP-T46G IP Phone User A User B Proxy Server F1. INVITE B F4. 180 Ringing F6. 200 OK F7. ACK F2. INVITE B F3. 180 Ringing F5. 200 OK F8. ACK Session1 established between User A and User B is active F9. INVITE(sendonly) Initiate three party conference F10. INVITE (sendonly) F11. 200 OK F12. 200 OK F13. ACK F14. 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.
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-T46G IP Phone Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server. The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
Appendix Step Action C Description sends the SIP INVITE request to User C. User C sends a SIP 180 Ringing F17 180 Ringing—User C to Proxy response to the proxy server. The 180 Server Ringing response indicates that the user is being alerted. The proxy server forwards the 180 F18 180 Ringing—Proxy Server to Ringing response to User A. User A User A hears the ring-back tone indicating that User C is being alerted.
Administrator’s Guide for SIP-T46G IP Phone 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.
Appendix dialplan.replace.prefix.X = dialplan.replace.replace.X = dialplan.replace.line_id.X = #Time Settings local_time.time_zone = local_time.time_zone_name = local_time.ntp_server1 = local_time.ntp_server2 = local_time.interval = local_time.dhcp_time = #Use the following parameters to set the time and date manually. local_time.manual_time_enable = local_time.date_format = local_time.time_format = #Auto DST Settings local_time.summer_time = local_time.dst_time_type = local_time.start_time = local_time.
Administrator’s Guide for SIP-T46G IP Phone sip.rfc2543_hold = #Hotline features.hotline_number = features.hotline_delay = #Web Server Type wui.http_enable = network.port.http = wui.https_enable = network.port.https = #DTMF Suppression features.dtmf.hide = features.dtmf.hide_delay = #Call Forward # In Phone Mode features.fwd_mode = 0 forward.always.enable = forward.always.target = forward.always.on_code = forward.always.off_code = forward.busy.enable = forward.busy.target = forward.busy.
Appendix account.1.timeout_fwd.target = account.1.timeout_fwd.timeout = account.1.timeout_fwd.on_code = account.1.timeout_fwd.off_code = #Call Transfer transfer.semi_attend_tran_enable = transfer.blind_tran_on_hook_enable = transfer.on_hook_trans_enable = transfer.tran_others_after_conf_enable = #Call Conference account.1.conf_type = account.1.conf_uri = #DTMF account.1.dtmf.type = account.1.dtmf.dtmf_payload = account.1.dtmf.info_type = #Distinctive Ring Tones account.1.
Administrator’s Guide for SIP-T46G IP Phone ldap.number_filter = ldap.host = 0.0.0.0 ldap.port = 389 ldap.base = ldap.user = ldap.password = ldap.max_hits = ldap.name_attr = ldap.numb_attr = ldap.display_name = ldap.version = ldap.search_delay = 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.
Appendix action_url.forward_incoming_call = action_url.reject_incoming_call = action_url.answer_new_incoming_call = action_url.transfer_finished = action_url.transfer_failed = #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.url = wallpaper_upload.
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Index Numeric C 180 Ring Workaround 75 Call Completion 802.
Administrator’s Guide for SIP-T46G IP Phone G N Getting Information from Status Indicators 221 NAT Traversal Getting Started Network Address Translation (NAT) 7 Group Call Pickup 92 155 Network Conference 155 89 No Answer Forward 82 H H.
Index SRTP 198 STUN Server 155 Suppressing the Display of DTMF Digits 105 T Table of Contents Time and Date vii 40 Transfer on Conference Hang Up Transfer via DTMF 109 Transport Layer Security (TLS) Troubleshooting 90 193 217 Troubleshooting Methods 217 Troubleshooting Solutions 222 TR-069 Device Management 174 U Upgrading Firmware 181 Use Outbound Proxy in Dialog User Agent Client (UAC) 2 User Agent Server (UAS) 3 User Password 76 34 V Verifying Startup Viewing Log Files VLAN
