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Summary of content (417 pages)

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Copyright © 2014 YEALINK NETWORK TECHNOLOGY Copyright © 2014 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.
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Note: This device is tested and complies with the limits for a Class B digital device, pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications.
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Yealink CP860 IP conference phone firmware contains third-party software under the GNU General Public License (GPL). Yealink uses software under the specific terms of the GPL. Please refer to the GPL for the exact terms and conditions of the license. The original GPL license, source code of components licensed under GPL and used in Yealink products can be downloaded online: http://www.yealink.com/GPLOpenSource.aspx?BaseInfoCateId=293&NewsCateId=293&CateId=293.
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About This Guide The guide is intended for administrators who need to properly configure, customize, manage, and troubleshoot the IP phone system rather than the end-users. It provides details on the functionality and configuration of CP860 IP conference phones. Many of the features described in this guide involve network settings, which could affect the IP phone’s performance in the network. So an understanding of the IP networking and prior knowledge of IP telephony concepts are necessary.
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Administrator’s Guide for CP860 IP conference phones  Chapter 5, “Configuring Audio Features” describes how to configure audio features on IP phones.  Chapter 6, “Configuring Security Features” describes how to configure security features on IP phones.  Chapter 7, “Resource Files” describes the resource files that can be downloaded by IP phones.  Chapter 8, “Troubleshooting” describes how to troubleshoot IP phones and provides some common troubleshooting solutions.
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Table of Contents About This Guide ...................................................................... v Documentations ............................................................................................................................... v In This Guide .................................................................................................................................... v Table of Contents ....................................................................
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Administrator’s Guide for CP860 IP conference phones Configuring Basic Features .................................................... 39 Contrast .......................................................................................................................................... 40 Backlight ......................................................................................................................................... 42 Web Server Type..........................................................
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Table of Contents Transfer on Conference Hang Up .............................................................................................. 152 Directed Call Pickup .................................................................................................................... 153 Group Call Pickup........................................................................................................................ 157 Call Return ...................................................................
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Administrator’s Guide for CP860 IP conference phones Voice Activity Detection ....................................................................................................... 299 Comfort Noise Generation .................................................................................................. 301 Jitter Buffer ............................................................................................................................ 302 Configuring Security Features .......................
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Table of Contents What do “on code” and “off code” mean? ....................................................................... 347 How to solve the IP conflict problem? ................................................................................ 347 How to reset your phone to factory configurations? ......................................................... 347 How to restore the administrator password? .................................................................... 348 Appendix ................
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Product Overview This chapter contains the following information about CP860 IP conference phones:  VoIP Principle  SIP Components  Introduction of CP860 IP Conference 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.
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Administrator’s Guide for CP860 IP conference phones SIP provides capabilities to:  Determine the location of the target endpoint -- SIP supports address resolution, name mapping, and call redirection.  Determine the media capabilities of the target endpoint -- Via Session Description Protocol (SDP), SIP determines the “lowest level” of common services between endpoints. Conferences are established using only the media capabilities that can be supported by all endpoints.
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Product Overview preferential to use this method when not using an application layer firewall. Application layer firewalls like to know what applications are flowing though which ports and it is possible to use content types of other applications other than the one you are trying to let through what has been denied. User agent server (UAS) UAS is a server that hosts the application responsible for receiving the SIP requests from a UAC, and on reception it returns a response to the request back to the UAC.
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Administrator’s Guide for CP860 IP conference phones This section lists the available physical features of CP860 IP conference phones. CP860 IP conference phone Physical Features: - 192 x 64 graphic LCD - One VoIP account - HD Voice: HD Codec - 1 mobile phone/PC port: 3.5mm - 1xRJ45 10/100Mbps Ethernet port - 2xEX mic ports - 1xUSB2.0 port - Security lock port - 3 LED indicators - Power adapter (optional): AC 100~240V input and DC 5V/2A output - Power over Ethernet (IEEE 802.
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Product Overview - Basic Features: DND, auto redial, live dialpad, dial plan, hotline, caller identity, auto answer. - Advanced Features: server redundancy, distinctive ring tones, remote phone book, LDAP, 802.1X authentication.     Codecs and Voice Features - Codecs: G.722, PCMU, PCMA, G.729, G.723, G.
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Administrator’s Guide for CP860 IP conference phones 6
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Getting Started This chapter provides basic information and installation instructions of CP860 IP conference phones. This chapter provides the following sections:  Connecting the IP Phone  Initialization Process Overview  Verifying Startup  Reading Icons  Configuration Methods  Provisioning Server  Configuring Basic Network Parameters  Upgrading Firmware This section introduces how to install CP860 IP conference phones with the components in packaging contents. 1.
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Administrator’s Guide for CP860 IP conference phones AC Power (Optional) To connect the AC power and network: 1. Connect the DC plug of the power adapter to the DC5V port on IP phones and connect the other end of the power adapter into an electrical power outlet. 2. Connect the included or a standard Ethernet cable between the Internet port on IP phones and the one on the wall or switch/hub device port.
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Getting Started To connect the PoE: 1. Connect the Ethernet cable between the Internet port on the IP phone and an available port on the in-line power switch/hub. Note If in-line power switch/hub is provided, you don’t need to connect the phone to the power adapter. Make sure the switch/hub is PoE-compliant. Important! Do not unplug or remove power to the phone while it is updating firmware and configurations.
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Administrator’s Guide for CP860 IP conference phones To connect the extension microphones: 1. Connect the free end of the optional extension microphone cable to one of the MIC ports on the phone. You can connect a USB flash drive to record and play back calls. To connect a USB flash drive: 1. Insert a USB flash drive into the USB port on the phone. You can connect a PC or mobile device to listen to the PC or mobile audio using your conference phone.
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Getting Started To connect a PC or mobile device: 1. Connect one end of the 3.5mm jack cable to the PC/mobile port on the phone, and connect the other end to the headset jack on the mobile device or the AUX/MIC jack on the PC. The initialization process of IP phones is responsible for network connectivity and operation of 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.
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Administrator’s Guide for CP860 IP conference phones Querying the DHCP (Dynamic Host Configuration Protocol) Server IP phones are capable of querying a DHCP server. DHCP is enabled on IP phones by default.
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Getting Started After connected to the power and network, the IP phone begins the initializing process by cycling through the following steps: 1. Three LED indicators on the phone illuminate solid red. 2. The message “Initializing…please wait” appears on the LCD screen when the IP phone starts up. 3. 4.
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Administrator’s Guide for CP860 IP conference phones Icon Description Call Hold Call Mute Ringer volume is 0 Keypad Lock Alphanumeric input mode Numeric input mode Multi-lingual lowercase letters input mode Multi-lingual uppercase letters input mode Multi-lingual uppercase and lowercase letters input mode Call Forward/Forwarded Calls Missed Calls Received Calls Placed Calls USB flash drive is inserted USB flash drive is detecting High Definition Voice IP phones can be configured automatically through con
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Getting Started  Web User Interface  Configuration Files An administrator or a user can configure and use 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 phone user interface. For more information, refer to Yealink_CP860_User_Guide, available online: http://www.yealink.com/DocumentDownload.
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Administrator’s Guide for CP860 IP conference phones configuration files (y000000000037.cfg and .cfg). You can use a text-based editing application to edit configuration files, and then store configuration files to a provisioning server. For more information on the provisioning server, refer to Provisioning Server on page 16. IP phones can obtain the provisioning server address during startup.
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Getting Started The provisioning server can be on the local LAN or anywhere on the Internet. Use the following procedure as a recommendation if this is your first provisioning server setup. For more information on how to set up a provisioning server, refer to Yealink_SIP-T2_Series_T19P_T4_Series_CP860_IP_Phones_Auto_Provisioning_Guide. To set up the provisioning server: 1. Install a provisioning server application or locate a suitable existing server. 2. Create an account and home directory. 3.
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Administrator’s Guide for CP860 IP conference phones 2. Create new common configuration files by performing the following steps: a) Create y000000000037.cfg files by using the Common CFG file from the distribution as templates. b) Edit the parameters in the file as desired. 3. Copy configuration files to the home directory of the provisioning server. 4. Reboot IP phones to trigger the auto provisioning process.
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Getting Started phones comply with the DHCP specifications documented in RFC 2131. If DHCP is used, IP phones connected to the network become operational without having to be manually assigned IP addresses and additional network parameters. Static DNS address(es) can be configured and used when DHCP is enabled. DHCP Option DHCP provides a framework for passing information to TCP/IP network devices.
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Administrator’s Guide for CP860 IP conference phones Parameter Vendor Class Identifier TFTP Server Name DHCP Option 60 Description Identify the vendor type. Identify a TFTP server when the 'sname' field 66 in the DHCP header has been used for DHCP options. Identify a bootfile when the 'file' field in the Bootfile Name 67 DHCP header has been used for DHCP options. For more information on DHCP options, refer to http://www.ietf.org/rfc/rfc2131.txt?number=2131 or http://www.ietf.org/rfc/rfc2132.
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Getting Started Details of Configuration Parameters: Parameters Permitted Values Default 0 or 2 0 network.internet_port.type Description: Configures the Internet (WAN) port type for IPv4 when the IP address mode is configured as IPv4 or IPv4&IPv6. 0-DHCP 2-Static IP Address Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the primary IPv4 DNS server when the static IPv4 DNS is enabled. Example: network.primary_dns = 202.101.103.55 Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Getting Started 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 settings will take effect after reboot. 4. Click OK to reboot the phone. To configure static DNS address when DHCP is used via web user interface: 1. Click on Network->Basic. 2. In the IPv4 Config block, mark the DHCP radio box. 3. Mark the On radio box in the Static DNS field.
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Administrator’s Guide for CP860 IP conference phones 4. Enter the desired values in the Primary DNS and Secondary DNS fields. 5. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot. 6. Click OK to reboot the phone. To configure DHCP via phone user interface: 1. Press Menu->Settings->Advanced Settings (Default password: admin) ->Network->WAN Port->IPv4->DHCP IPv4 Client. 2. Press the Save soft key to accept the change.
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Getting Started Procedure Network parameters can be configured manually using the configuration files or locally. Configure network parameters of the IP phone manually. Parameters: network.internet_port.type Configuration File .cfg network.ip_address_mode network.internet_port.ip network.internet_port.mask network.internet_port.gateway network.primary_dns network.secondary_dns Configure network parameters of the IP phone manually.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the IP address mode. 0-IPv4 1-IPv6 2-IPv4&IPv6 Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Basic->Internet Port->Mode (IPv4/IPv6) Phone User Interface: Menu->Settings->Advanced Settings (Default password: admin) ->Network->WAN Port ->IP Mode network.internet_port.
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Getting Started Parameters Permitted Values Default effect. Web User Interface: Network->Basic->IPv4 Config->Static IP Address->Subnet Mask Phone User Interface: Menu->Settings->Advanced Settings (Default password: admin) ->Network->WAN Port ->IPv4->Static IPv4 Client->Subnet Mask network.internet_port.
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Administrator’s Guide for CP860 IP conference phones Parameters network.secondary_dns Permitted Values Default IPv4 Address Blank Description: Configures the secondary IPv4 DNS server when the IP address mode is configured as IPv4 or IPv4&IPv6, and the Internet (WAN) port type for IPv4 is configured as Static IP Address. Example: network.secondary_dns = 202.101.103.54 Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Getting Started 4. Click OK to reboot the phone. To configure a static IPv4 address via web user interface: 1. Click on Network->Basic. 2. In the IPv4 Config block, mark the Static IP Address radio box. 3. Enter the 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 settings will take effect after reboot. 5. Click OK to reboot the phone.
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Administrator’s Guide for CP860 IP conference phones 4. Enter the desired values in the IPv4 Address, Subnet Mask, Default Gateway, Primary DNS and Secondary DNS fields respectively. 5. Press the Save soft key to accept the change. The IP phone reboots automatically to make settings effective after a period of time. Note Wrong network settings may result in inaccessibility of your phone and may also have an impact on your network performance.
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Getting Started Full-duplex Full-duplex transmission refers to transmitting voice or data in both directions at the same time; this means one device can send data on the line while receiving data. You can configure the full-duplex transmission on Internet port for IP phones to transmit in 10Mbps or 100Mbps. Procedure The transmission method of Internet port can be configured using the configuration files or locally. Configure the transmission methods Configuration File y000000000037.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Network->Advanced->Port Link->WAN Port Link Phone User Interface: None To configure the transmission method of Ethernet port via web user interface: 1. Click on Network->Advanced. 2. Select the desired value from the pull-down list of WAN Port Link. 3. Click Confirm to accept the change. This section provides information on upgrading the IP phone firmware.
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Getting Started Upgrade via Web User Interface To manually upgrade firmware via web user interface, you need to store the firmware to the local system in advance. To upgrade firmware manually via web user interface: 1. Click on Settings->Upgrade. 2. Click Browse. 3. Locate the firmware from the local system. 4. Click Upgrade. A dialog box pops up to prompt “Firmware of the SIP phone will be updated. It will take 5 minutes to complete. Please don't power off!”. 5.
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Administrator’s Guide for CP860 IP conference phones Procedure Configuration changes can be performed using the configuration files or locally. Configure the way for the IP phone to check for configuration files. Parameters： auto_provision.power_on auto_provision.repeat.enable auto_provision.repeat.minutes Configuration File y000000000037.cfg auto_provision.weekly.enable auto_provision.weekly.begin_time auto_provision.weekly.end_time auto_provision.weekly.dayofweek Specify the access URL of firmware.
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Getting Started Parameters Permitted Values Default Description: Enables or disables the IP phone to perform an auto provisioning process repeatedly. 0-Disabled 1-Enabled Web User Interface: Settings->Auto provision->Repeatedly Phone User Interface: None auto_provision.repeat.minutes Integer from 1 to 43200 1440 Description: Configures the interval (in minutes) for the IP phone to perform an auto provisioning process repeatedly. Note: It works only if the parameter “auto_provision.repeat.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the begin time of the day for the IP phone to perform an auto provisioning process weekly. Note: It works only if the parameter “auto_provision.weekly.enable” is set to 1(Enabled). Web User Interface: Settings->Auto provision->Time Phone User Interface: None auto_provision.weekly.
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Getting Started Parameters Permitted Values Default provisioning process every Sunday and Monday. Note: It works only if the parameter “auto_provision.weekly.enable” is set to 1(Enabled). Web User Interface: Settings->Auto provision->Day of week Phone User Interface: None firmware.url URL within 511 characters Blank Description: Configures the access URL of the firmware file. Example： firmware.url = http://192.168.1.20/2.71.0.140.
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Administrator’s Guide for CP860 IP conference phones 2. Make the desired change. 3. Click Confirm to accept the change. When the “Power On” is set to On, the IP phone will check configuration files stored on the provisioning server during startup and then will download firmware from the server.
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Configuring Basic Features This chapter provides information for making configuration changes for the following basic features:  Contrast  Backlight  Web Server Type  User Password  Administrator Password  Phone Lock  Time and Date  Language  Logo Customization  Softkey Layout  Key as Send  Dial Plan  Hotline  Directory  Search Source List in Dialing  Call Log  Missed Call Log  Local Directory  Live Dialpad  Call Waiting  Auto Redial  Auto An
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Administrator’s Guide for CP860 IP conference phones  Use Outbound Proxy in Dialog  SIP Session Timer  Session Timer  Call Hold  Call Forward  Call Transfer  Network Conference  Transfer on Conference Hang Up  Directed Call Pickup  Group Call Pickup  Call Return  Calling Line Identification Presentation  Connected Line Identification Presentation  DTMF  Suppress DTMF Display  Transfer via DTMF  Intercom Contrast determines the readability of the texts disp
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Configuring Basic Features Phone User Interface Configure the contrast of the LCD screen. Details of Configuration Parameters: Parameters phone_setting.contrast Permitted Values Default Integer from 1 to 10 6 Description: Configures the contrast of the LCD screen. Note: We recommend that you set the contrast of the LCD screen to 6 as a more comfortable level.
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Administrator’s Guide for CP860 IP conference phones Backlight determines the brightness of the LCD screen display, allowing users to read easily in dark environments. Backlight time specifies the delay time to turn off the backlight when the IP phone is inactive. You can configure the backlight time as one of the following types:  Always On: Backlight is turned on permanently.
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Configuring Basic Features 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 Time (seconds). 3. Click Confirm to accept the change. To configure the backlight via phone user interface: 1. Press Menu->Settings->Basic Settings->Display->Backlight Settings. 2. Press the 3. Press the Save soft key to accept the change. or soft key to select the desired value from the Backlight Time field.
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Administrator’s Guide for CP860 IP conference phones Specify the web access type, HTTP port and HTTPS port. Web User Interface Local Navigate to: http:///servl et?p=network-adv&q=load Phone User Interface Specify the web access type. Details of Configuration Parameters: Parameters wui.http_enable Permitted Values Default 0 or 1 1 Description: Enables or disables the IP phone to access its web user interface using HTTP protocol.
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Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables the IP phone to access its web user interface using HTTPS protocol. 0-Disabled 1-Enabled Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network-> Advanced-> Web Server->HTTPS Phone User Interface: Menu->Settings->Advanced Settings (Default password: admin) ->Network-> Webserver Type-> HTTPS Status network.port.
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Administrator’s Guide for CP860 IP conference phones The default HTTPS port is 443. 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 phone. To configure the web server type via phone user interface: 1. Press Menu->Settings->Advanced Settings (Default password: admin) ->Network->Webserver Type. 2. Press the or 3. Enter the HTTP port in the HTTP Port field. 4. Press the 5.
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Configuring Basic Features Procedure User password can be changed using the configuration files or locally. Change the user password of the Configuration File IP phone. y000000000037.cfg Parameter: security.user_password Change the user password of the IP phone. Local Web User Interface Navigate to: http:///servlet ?p=security&q=load Details of the Configuration Parameter: Parameter security.
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Administrator’s Guide for CP860 IP conference phones Valid characters are ASCII characters 32-126(0x20-0x7E) except 58(3A). 4. Note Click Confirm to accept the change. If logging into the web user interface of the IP phone with the user credential, the user needs to enter the current user password in the Old Password field. Advanced menu options are strictly used by administrators. Users can configure them only if they have administrator privileges.
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Configuring Basic Features Details of the Configuration Parameter: Parameter security.user_password Permitted Values String within 32 characters Default admin Description: Configures the password of the administrator for web server access. The IP phone uses “admin” as the default administrator password. Example: security.user_password = admin:password123 means setting the password of administrator (current user name is “admin”) to password123.
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Administrator’s Guide for CP860 IP conference phones 3. Enter a new administrator password in the New PWD field and Confirm PWD field. Valid characters are ASCII characters 32-126(0x20-0x7E). 4. Press the Save soft key to accept the change. Phone lock is used to lock the IP phone to prevent it from unauthorized use. Once the IP phone is locked, a user must enter the password to unlock it. IP phones offer three types of phone lock: Menu Key, Function Keys and All Keys.
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Configuring Basic Features ures-phonelock&q=load Assign a keypad lock key. Navigate to: http:///servlet?p=dssk ey&model=2&q=load Phone User Configure the phone lock type. Interface Configure the unlock PIN. Details of Configuration Parameters: Parameters phone_setting.phone_lock.enable Permitted Values Default 0 or 1 0 Description: Enables or disables phone lock feature.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default characters within 15 digits 123 Lock->Keypad Lock Type phone_setting.phone_lock.unlock_pin Description: Configures the password for unlocking the keypad. Web User Interface: Features->Phone Lock->Phone Unlock PIN (0~15 Digit) Phone User Interface: Menu->Settings->Basic Settings->Phone Unlock PIN phone_setting.phone_lock.
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Configuring Basic Features 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. Select the desired type from the pull-down list of Keypad Lock Type. 4. Enter unlock PIN (numeric characters) in the phone Unlock PIN (0~15 Digit) field. 5. Enter the desired time in the phone Lock Time Out (0~3600s) field. 6. Click Confirm to accept the change.
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Administrator’s Guide for CP860 IP conference phones 3. Press the or soft key to select the desired type from the Lock type field. 4. Press the Save soft key to accept the change. To configure the unlock PIN via phone user interface: 1. Press Menu->Settings->Basic Settings->Phone Unlock PIN. 2. Enter the current unlock PIN in the Current PIN field. 3. Enter the new unlock PIN in the New PIN field. 4. Enter the new unlock PIN again in the Confirm PIN field. 5.
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Configuring Basic Features Option Methods of Configuration Configuration Files Time Format Web User Interface Phone User Interface Web User Interface Date Phone User Interface Configuration Files Date Format Web User Interface Phone User Interface Configuration Files Daylight Saving Time Web User Interface Procedure Configuration changes can be performed using the configuration files or locally. Configure NTP by DHCP priority feature and DHCP time features. Parameters: local_time.
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Administrator’s Guide for CP860 IP conference phones formats. Parameters: local_time.time_format local_time.date_format Configure NTP by DHCP priority feature. Configure the NTP server, time zone and DST. Configure the time and date Web User Interface manually. Configure the time and date formats. Navigate to: Local http:///servlet ?p=settings-datetime&q=load Configure the NTP server and time zone. Phone User Interface Configure the time and date manually.
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Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables the IP phone to update time with the offset time obtained from the DHCP server. 0-Disabled 1-Enabled Note: It is only available to offset from GMT 0. Web User Interface: Settings->Time & Date->DHCP Time Phone User Interface: Menu->Settings->Basic Settings->Time & Date->DHCP Time local_time.ntp_server1 IP Address or Domain Name cn.pool.ntp.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the interval (in seconds) to update time and date from the NTP server. Example: local_time.interval = 1000 Web User Interface: Settings->Time & Date->Synchronism (15~86400s) Phone User Interface: None local_time.time_zone -11 to +13 +8 Description: Configures the time zone. For more available time zones, refer to Appendix B: Time Zones on page 351. Example: local_time.
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Configuring Basic Features Parameters Permitted Values Default Description: Configures Daylight Saving Time (DST) feature. 0-Disabled 1-Enabled 2-Automatic Web User Interface: Settings->Time & Date->Daylight Saving Time Phone User Interface: Menu->Settings->Basic Settings->Time & Date->SNTP Settings->Daylight Saving local_time.dst_time_type 0 or 1 0 Description: Configures the DST time type. 0-By Date 1-By Week Note: It works only if the parameter “local_time.summer_time” is set to 1 (Enabled).
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Hour of Day: 0=1am, 1=2am,…, 23=12pm Note: It works only if the parameter “local_time.summer_time” is set to 1 (Enabled). Web User Interface: For DST By Date: Settings-> Time & Date->Start Date For DST By Week: Settings-> Time & Date->DST Start Month/DST Start Day of Week/DST Start Day of Week Last in Month/ Start Hour of Day Phone User Interface: None local_time.
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Configuring Basic Features Parameters local_time.offset_time Permitted Values Default Integer from -300 to 300 Blank Description: Configures the offset time (in minutes) of DST. Note: It works only if the parameter “local_time.summer_time” is set to 1 (Enabled). Web User Interface: Settings->Time & Date->Offset (minutes) Phone User Interface: None local_time.manual_time_enable 0 or 1 0 Description: Configures the IP phone to obtain time from the NTP server or manual settings.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the date format. 0-WWW MMM DD 1-DD-MMM-YY 2-YYYY-MM-DD 3-DD/MM/YYYY 4-MM/DD/YY 5-DD MMM YYYY 6-WWW DD MMM Web User Interface: Settings->Time & Date->Date Format Phone User Interface: Menu->Settings->Basic Settings->Time & Date->Time & Date Format->Date Format To configure NTP by DHCP priority feature via web user interface: 1. Click on Settings->Time & Date. 2.
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Configuring Basic Features 4. Enter the domain names or IP addresses in the Primary Server and Secondary Server fields respectively. 5. Enter the desired time interval in the Synchronism (15~86400s) field. 6. Select the desired value from the pull-down list of Daylight Saving Time. If you select Enabled, do one of the following: - Mark the DST By Date radio box in the Fixed Type field. Enter the start time in the Start Date field. Enter the end time in the End Date field.
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Administrator’s Guide for CP860 IP conference phones Enter the desired time in the End Hour of Day field. 7. Enter the desired offset time in the Offset (minutes) field. 8. Click Confirm to accept the change. To configure the time and date manually via web user interface: 1. Click on Settings->Time & Date. 2. Select Enabled from the pull-down list of Manual Time. 3. Enter the time and date in the corresponding fields. 4. Click Confirm to accept the change.
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Configuring Basic Features 2. Select the desired value from the pull-down list of Time Format. 3. Select the desired value from the pull-down list of Date Format. 4. Click Confirm to accept the change. To configure the NTP server and time zone via phone user interface: 1. Press Menu->Settings->Basic Settings->Time & Date->SNTP Settings. 2. Press the or soft key to select the time zone that applies to your area from the Time Zone field. The default time zone is "+8 China(Beijing)". 3.
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Administrator’s Guide for CP860 IP conference phones IP phones support multiple languages. Languages used on the phone user interface and web user interface can be specified respectively as required. The following table lists the languages supported by the phone user interface and the web user interface respectively.
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Configuring Basic Features Available Language Associated Language Pack Spanish lang-Spanish.txt Turkish lang-Turkish.txt Russian lang-Russian.txt To update translation of a built-in language, the file name of the language file cannot be changed. For more information, refer to Yealink_SIP-T2_Series_T19P_T4_Series_CP860_IP_Phones_Auto_Provisioning_Guide. Procedure Loading language pack can only be performed using the configuration files.
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Administrator’s Guide for CP860 IP conference phones the language is not supported by the IP phone, the web user interface uses English). You can specify the languages for the phone user interface and web user interface respectively. Procedure Specify the language for the web user interface or the phone user interface using the configuration files or locally. Specify the languages for the phone user interface and the web user interface. Configuration File y000000000037.cfg Parameters: lang.gui lang.
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Configuring Basic Features Parameters Permitted Values Default Description: Configures the language used on the web user interface. Example: lang.wui = English Permitted Values: English, Chinese_S, Chinese_T, German, French, Turkish, Italian, Polish, Spanish, Russian or Portuguese Note: If the language of your browser is not supported by the IP phone, the web user interface will use English by default.
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Administrator’s Guide for CP860 IP conference phones Logo customization allows unifying the IP phone appearance or displaying a custom image on the idle screen such as a company logo, instead of the default system logo. The logo file format must be *.dob, and the resolution of the LCD screen is 192*64 graphic. Note Before uploading your custom logo to IP phones, ensure the logo file is in the correct format.
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Configuring Basic Features Parameters Permitted Values Default URL within 511 characters Blank to upload a custom logo file to the IP phone). Web User Interface: Features->General Information->Use Logo Phone User Interface: None lcd_logo.url Description: Configures the access URL of the custom logo file. Example： The following example uses HTTP to download the custom logo file (logo.dob) from the provisioning server 192.168.10.25. lcd_logo.url = http://192.168.10.25/logo.
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Administrator’s Guide for CP860 IP conference phones 3. Click Browse to select the logo file from your local system. 4. Click Upload to upload the file. 5. Click Confirm to accept the change. The custom logo screen and the idle screen are displayed alternately. Softkey layout is used to customize the soft keys at the bottom of the LCD screen to best meet users’ requirements. In addition to specifying which soft keys to display, you can determine their display order.
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Configuring Basic Features Call State Default Soft Key Optional Soft Key DPickup RingBack Empty Empty Empty Switch Empty Cancel RingBack SemiAttendTransBack Transfer Empty Empty Switch Empty Cancel Transfer Empty Hold Mute Conference SWAP Cancel NewCall Switch Talk Answer Reject Start Record Pause Record Resume Record Stop Record Talking Hold Transfer Empty Resume Switch NewCall Answer Cancel Reject Start Record Pause Record Resume Record Stop Record Held Empty Empty E
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Administrator’s Guide for CP860 IP conference phones Call State Default Soft Key Optional Soft Key Resume Record Stop Record PreTrans Transfer Empty IME Directory Delete Switch Cancel Send Empty Empty Hold Switch Split Answer Cancel Reject Mute Conferenced Manager Start Record Pause Record Resume Record Stop Record Procedure Softkey layout can be configured using the configuration files or locally. Specify the access URL of the softkey layout template.
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Configuring Basic Features Details of Configuration Parameters: Parameters custom_softkey_call_failed.url Permitted Values Default URL within 511 characters Blank Description: Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Call Failed state. Example: The following example uses HTTP to download the CallFailed state file from the “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port. custom_softkey_call_failed.url = http://10.2.8.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default URL within 511 characters Blank Web User Interface: None Phone User Interface: None custom_softkey_dialing.url Description: Configures the access URL of the custom file for the soft key presented on the LCD screen when in the Dialing state. Example: The following example uses HTTP to download the Dialing state file from the “XMLfiles” directory on provisioning server 10.2.8.16 using 8080 port.
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Configuring Basic Features Parameters Permitted Values Default Example: The following example uses HTTP to download the Talking state file from the “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 Web User Interface: None Phone User Interface: None To configure softkey layout via web user interface: 1. Click on Settings->Softkey Layout. 2. Select the desired value from the pull-down list of Custom Softkey.
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Administrator’s Guide for CP860 IP conference phones only if Key tone is enabled. Procedure Key as send can be configured using the configuration files or locally. Configure a send key. Parameter: features.key_as_send Configuration File y000000000037.cfg Configure a key tone and send tone. Parameters: features.key_tone eatures.send_key_tone Configure a send key. Navigate to: http:///servlet ?p=features-general&q=load Web User Interface Local Configure a key tone and send tone.
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Configuring Basic Features Parameters Permitted Values Default 0 or 1 1 Phone User Interface: Menu->Features->Key as Send features.key_tone Description: Enables or disables the IP phone to play a tone when a user presses a key on your phone keypad. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will play a tone when a user presses a key on your phone keypad. Web User Interface: Features->Audio->Key Tone Phone User Interface: None features.
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Administrator’s Guide for CP860 IP conference phones 2. Select the desired value from the pull-down list of Key As Send. 3. Click Confirm to accept the change. To configure a key tone and send tone via web user interface: 1. Click on Features->Audio. 2. Select the desired value from the pull-down list of Key Tone. 3. Select the desired value from the pull-down list of Send Sound. 4. Click Confirm to accept the change. To configure key as send via phone user interface: 1.
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Configuring Basic Features 3. Note Press the Save soft key to accept the change. Send tone works only if key tone is enabled. Key tone is enabled by default. Regular expression, often called a pattern, is an expression that specifies a set of strings. A regular expression provides a concise and flexible means to “match” (specify and recognize) strings of text, such as particular characters, words, or patterns of characters.
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Administrator’s Guide for CP860 IP conference phones "([1-9])([2-7])3" would match “923”, “153”, “673”, etc. The “$” followed by the sequence number of a parenthesis means the characters placed in the parenthesis. The sequence number stands for the corresponding parenthesis. Example: A replace rule configuration, Prefix: "001(xxx)45(xx)", Replace: $ "9001$145$2". When you dial out "0012354599" on your phone, the IP phone will replace the number with "90012354599".
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Configuring Basic Features Parameters Permitted Values Default Description: Configures the entered number to be replaced. Example: dialplan.replace.prefix.1 = 00 Web User Interface: Settings->Dial Plan->Replace Rule->Prefix Phone User Interface: None dialplan.replace.replace.X (X ranges from 1 to 100) String within 32 characters Blank Description: Configures the alternate number to replace the entered number. Example: dialplan.replace.replace.
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Administrator’s Guide for CP860 IP conference phones 3. Enter the string in the Replace field. 4. 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, IP phones will automatically dial out the numbers without pressing the send key. IP phones support up to 100 dial-now rules, which can be created either one by one or in batch using a dial-now rule template.
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Configuring Basic Features Configure the access URL of the dial-now template. Parameters: dialplan_dialnow.url Create the dial-now rule for the IP phone. Navigate to: http:///servlet Local Web User Interface ?p=settings-dialnow&q=load Configure the delay time for the dial-now rule. Navigate to: http:///servlet ?p=features-general&q=load Details of Configuration Parameters: Parameters dialplan.dialnow.rule.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Features->General Information->Time-Out for Dial-Now Rule Phone User Interface: None dialplan_dialnow.url URL within 511 characters Description: Configures the access URL of the dial-now rule template file. Example: dialplan_dialnow.url = http://192.168.10.25/dialnow.xml Web User Interface: None Phone User Interface: None To create a dial-now rule via web user interface: 1. Click on Settings->Dial Plan->Dial-now.
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Configuring Basic Features 2. Enter the desired time within 1-14 (in seconds) in the Time-Out for Dial-Now Rule field. 3. Click Confirm to accept the change. Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country. When the entered numbers match the predefined area code rule, the IP phone will automatically add the area code before the numbers when dialing out them. IP phones only support one area code rule.
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Administrator’s Guide for CP860 IP conference phones Navigate to: http:///servlet ?p=settings-areacode&q=load Details of Configuration Parameters: Parameters dialplan.area_code.code Permitted Values Default String within 16 characters Blank Description: Configures the area code to be added before the entered numbers when dialing out. Example: dialplan.area_code.code = 010 Web User Interface: Settings->Dial Plan->Area Code->Code Phone User Interface: None dialplan.area_code.
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Configuring Basic Features 2. Enter desired values in the Code, Min Length (1-15) and Max Length (1-15) fields. 3. Click Confirm to accept the change. Block out rule prevents users from dialing out specific numbers. When the entered numbers match the predefined block out rule, the LCD screen prompts “Forbidden Number”. IP phones support up to 10 block out rules. Procedure Block out rule can be created using the configuration files or locally.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the block out numbers. Example: dialplan.block_out.number.1 = 1234 Web User Interface: Settings->Dial Plan->Block Out->BlockOut NumberX Phone User Interface: None To create a block out rule via web user interface: 1. Click on Settings->Dial Plan->Block Out. 2. Enter the desired value in the BlockOut Number field. 3. Click Confirm to add the block out rule.
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Configuring Basic Features Procedure Hotline can be configured using the configuration files or locally. Configure the hotline number. Parameter: features.hotline_number Configuration File Specify the time (in seconds) the y000000000037.cfg IP phone waits to automatically dial out the hotline number. Parameter: features.hotline_delay Configure the hotline number. Specify the time (in seconds) the IP phone waits to automatically Web User Interface dial out the hotline number.
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Administrator’s Guide for CP860 IP conference phones Parameter Permitted Values Default Description: Configures the waiting time (in seconds) for the IP phone to automatically dial out the hotline number. If it is set to 0 (0s), the IP phone will immediately dial out the preconfigured hotline number when you press the off-hook key. If it is set to a value greater than 0, the IP phone will wait the designated seconds before dialing out the predefined hotline number when you press the off-hook key.
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Configuring Basic Features Directory provides easy access to frequently used lists. The lists can be Local Directory, History, Remote Phone Book and LDAP. The desired list(s) can be added to Directory using a directory file. For more information on the directory file, refer to Directory Template on page 327. Procedure Directory can be configured using the configuration files or locally. Specify the access URL of the Configuration File Directory file. y000000000037.cfg Parameter: directory_setting.
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Administrator’s Guide for CP860 IP conference phones click 5. To adjust the display order of enabled lists, select the desired list and then click or 6. . . Click Confirm to accept the change. The IP phone LCD screen will display the enabled list(s) in the adjusted order. Search source list in dialing allows the IP phone to automatically search entries from the search source list based on the entered string, and display results on the dialing screen.
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Configuring Basic Features ?p=contacts-favorite&q=load Details of the Configuration Parameter: Parameter super_search.url Permitted Values Default URL within 511 characters Blank Description: Configures the access URL of the super search template. Web User Interface: Directory->Setting->Search Source List In Dialing Phone User Interface: None To configure search source list in dialing via web user interface: 1. Click on Directory->Setting. 2.
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Administrator’s Guide for CP860 IP conference phones The dialing screen displays the search results in the adjusted order. Call log contains call information such as remote party identification, time and date, and call duration. IP phones maintain a local call log. Call log consists of four lists: Missed calls, Placed calls, Received calls and Forwarded calls. Each call log list supports up to 100 entries. To store call information, you must enable the save call log feature in advance.
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Configuring Basic Features 2. Select the desired value from the pull-down list of Save Call Log. 3. Click Confirm to accept the change. To configure the call log via phone user interface: 1. Press Menu-> Features-> History Setting. 2. Press the 3. Press the Save soft key to accept the change. or soft key to select the desired value from the History Record field.
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Administrator’s Guide for CP860 IP conference phones http:///servlet ?p=account-basic&q=load&acc =0 Details of the Configuration Parameter: Parameter Permitted Values Default 0 or 1 1 account.X.missed_calllog (X =1) Description: Enables or disables the IP phone to record missed calls for account X. 0-Disabled 1-Enabled If it is set to 0 (Disabled), there is no indicator displaying on the LCD screen, the IP phone does not log the missed call in the Missed Calls list.
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Configuring Basic Features The IP phone maintains a local directory. The local directory can store up to 1000 contacts and 48 groups (including the default groups: Company, Family and Friend). When adding a contact to the local directory, in addition to name and phone numbers, you can also specify the ring tone and group for the contact. Contacts and groups can be added either one by one or in batch using a local contact file.
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Administrator’s Guide for CP860 IP conference phones 2. In the Group Setting block, enter the new group name in the Group field. 3. Select the desired group ring tone from the pull-down list of Ring. 4. Click Add to add the group. To add a contact to the local directory via web user interface: 1. Click on Directory->Local Directory. 2. Enter the name and the office, mobile or other numbers in the corresponding fields. 100 3. Select the desired ring tone from the pull-down list of Ring Tone. 4.
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Configuring Basic Features 5. Click Add to add the contact. To add a group to the local directory via phone user interface: 1. Press Menu->Directory->Local Directory. 2. Press the AddGrp soft key. 3. Enter the desired group name in the Name field. 4. Press the 5. Press the Save soft key to accept the change or the Back soft key to cancel. or soft key to select the desired ring tone from the Ring Tones field. To add a contact to the local directory via phone user interface: 1.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Enables or disables live dialpad feature. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will automatically dial out the entered phone number in the pre-dialing screen without pressing a send key. Web User Interface: Settings->Preference->Live Dialpad Phone User Interface: None phone_setting.
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Configuring Basic Features 3. Enter the desired delay time in the Inter Digit Time (1~14s) field. 4. Click Confirm to accept the change. Call waiting allows IP phones to receive a new incoming call when there is already an active call. The new incoming call is presented to the user visually on the LCD screen. Call waiting tone allows the IP phone to play a short tone, to remind the user audibly of a new incoming call during conversation. Call waiting tone works only if call waiting is enabled.
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Administrator’s Guide for CP860 IP conference phones Details of Configuration Parameters: Parameters call_waiting.enable Permitted Values Default 0 or 1 1 Description: Enables or disables call waiting feature. 0-Disabled 1-Enabled If it is set to 0 (Disabled), a new incoming call is automatically rejected by the IP phone with a busy message while during a call. If it is set to 1 (Enabled), the LCD screen will present a new incoming call while during a call.
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Configuring Basic Features Parameters Permitted Values Default Description: Configures the call waiting on code to activate the server-side call waiting feature. The IP phone will send the call waiting on code to the server when you activate call waiting feature on the IP phone. Example: call_waiting.on_code = *71 Web User Interface: Features->General Information->Call Waiting On Code Phone User Interface: Menu->Features->Call Waiting->On Code call_waiting.
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Administrator’s Guide for CP860 IP conference phones 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. Select the desired value from the pull-down list of Call Waiting Tone. 3. Click Confirm to accept the change. To configure call waiting and call waiting tone via phone user interface: 1. Press Menu->Features->Call Waiting. 2.
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Configuring Basic Features Auto redial allows IP phones to redial a busy number after the first attempt. Both the number of attempts and waiting time between redials are configurable. Procedure Auto redial can be configured using the configuration files or locally. Configure auto redial feature. Parameters: Configuration File y000000000037.cfg auto_redial.enable auto_redial.interval auto_redial.times Configure auto redial feature.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the interval (in seconds) for the IP phone to wait between redials. The IP phone redials the dialed number at regular intervals till the callee answers the call. Web User Interface: Features->General Information->Auto Redial Interval (1~300s) Phone User Interface: Menu->Features->Auto Redial->Redial Interval auto_redial.
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Configuring Basic Features 4. Enter the desired times in the Auto Redial Times (1~300) field. The default value is 10. 5. Click Confirm to accept the change. To configure auto redial via phone user interface: 1. Press Menu->Features->Auto Redial. 2. Press the 3. Enter the desired time in the Redial Interval field. or soft key to select Enable from the Auto Redial field. The default time interval is 10 seconds. 4. Enter the desired times in the Redial Times field. The default value is 10. 5.
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Administrator’s Guide for CP860 IP conference phones account.X.auto_answer_mute_enable Specify a period of delay time for auto y000000000037.cfg answer. Parameter: features.auto_answer_delay Configure auto answer. Navigate to: http:///servlet?p=a Web User Interface Local ccount-basic&q=load&acc=0 Specify a period of delay time for auto answer. http:///servlet?p=f eatures-general&q=load Phone User Interface Configure auto answer.
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Configuring Basic Features Parameters Permitted Values Default If it is set to 1 (Enabled), the IP phone can mute the local microphone when an incoming call is answered automatically. Web User Interface: Account->Basic->Auto Answer Mute Phone User Interface: Menu->Features->Auto Answer->Auto Answer Mute features.auto_answer_delay (X = 1) Integer from 1 to 4 1 Description: Configures the delay time (in seconds) before the IP phone automatically answers an incoming call.
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Administrator’s Guide for CP860 IP conference phones 2. Enter the desired time (in seconds) in the Auto-Answer Delay (1~4s) field. 3. Click Confirm to accept the change. To configure auto answer and auto answer mute via phone user interface: 1. Press Menu->Features->Auto Answer. 2. Press the or soft key to select Enable from the Auto Answer field. 3. Press the or soft key to select Enable from the Auto Answer Mute field. 4. Press the Save soft key to accept the change.
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Configuring Basic Features Max-Forwards: 70 User-Agent: Yealink CP860 37.72.0.2 Privacy: id Supported: replaces Allow-Events: talk,hold,conference,refer,check-sync P-Preferred-Identity: Content-Length: 302 The anonymous call on code and anonymous call off code configured on IP phones are used to activate/deactivate the server-side anonymous call feature. They may vary on different servers.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default instead of the caller’s identity. Web User Interface: Account->Basic->Local Anonymous Phone User Interface: Menu->Features->Anonymous Call->Local Anonymous account.X.send_anonymous_code 0 or 1 (X = 1) 0 Description: Configures the IP phone to send anonymous on/off code to activate/deactivate the server-side anonymous call feature.
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Configuring Basic Features Parameters Permitted Values Default String within 32 characters Blank Menu->Features->Anonymous Call->On Code account.X.anonymous_call_offcode (X = 1) Description: Configures the anonymous call off code to deactivate the server-side anonymous call feature. Example: account.1.anonymous_call_offcode = *87 Note: It works only if the parameter “account.X.send_anonymous_code” is set to 0 (Off Code).
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Administrator’s Guide for CP860 IP conference phones 1. Press Menu->Features->Anonymous Call. 2. Press the 3. (Optional.) Press the or soft key to select Enable from the Local Anonymous field. or soft key to select the desired value from the Anonymous Code field. 4. (Optional.) Enter the anonymous call on code in the On Code field. 5. (Optional.) Enter the anonymous call off code in the Off Code field. 6. Press the Save soft key to accept the change.
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Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables anonymous call rejection feature. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will automatically reject incoming calls from users enabled anonymous call feature. The anonymous user’s phone LCD screen presents “Anonymity Disallowed”. Web User Interface: Account->Basic->Anonymous Call Rejection Phone User Interface: Menu->Features->Anonymous Call->Anonymous Rejection account.X.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Web User Interface: Account->Basic->Anonymous Call Rejection->Off Code Phone User Interface: Menu->Features->Anonymous Call->Reject Off Code To configure anonymous call rejection via web user interface: 1. Click on Account->Basic. 2. Select the desired value from the pull-down list of Anonymous Call Rejection. 3. (Optional.) Enter the anonymous call rejection on code in the On Code field. 4. (Optional.
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Configuring Basic Features any of the IP phone’s registrations, the other registrations are not affected. For more information on call forward, refer to Call Forward on page 139. The DND on code and DND off code configured on IP phones are used to activate/deactivate the server-side DND feature. They may vary on different servers. Return Message When DND This feature defines the return code and the reason of the SIP response message for the rejected incoming call when DND is enabled on IP phones.
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Administrator’s Guide for CP860 IP conference phones Configure DND. Details of Configuration Parameters: Parameters features.dnd.enable Permitted Values Default 0 or 1 0 Description: Enables or disables DND feature. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will reject incoming calls on all accounts. Web User Interface: Features->Forward& DND->DND->DND Status Phone User Interface: Menu->Features->DND->DND Enable features.dnd.
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Configuring Basic Features Parameters Permitted Values Default 404, 480 or 486 480 Features->Forward& DND->DND->DND Off Code Phone User Interface: Menu->Features->DND->Off Code features.dnd_refuse_code Description: Configures a return code and reason of SIP response messages when rejecting an incoming call by DND. A specific reason is displayed on the caller’s phone LCD screen.
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Administrator’s Guide for CP860 IP conference phones 2. In the desired programable key field, select DND from the pull-down list of Type. 3. Click Confirm to accept the change. 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 DND Status field. 3. (Optional.) Enter the DND on code in the DND On Code field. 4. (Optional.) Enter the DND off code in the DND Off Code field. 5.
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Configuring Basic Features 2. Select the desired type from the pull-down list of Return Code When DND. 3. Click Confirm to accept the change. Busy tone is audible to the other party, indicating that the call connection has been broken when one party releases a call. Busy tone delay can define a period of time during which the busy tone is audible. Procedure Busy tone delay can be configured using the configuration files or locally. Configure the busy tone delay Configuration File y000000000037.
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Administrator’s Guide for CP860 IP conference phones Details of the Configuration Parameter: Parameter Permitted Values Default 0, 3 or 5 0 features.busy_tone_delay Description: Configures the duration time (in seconds) for the busy tone. When one party releases the call, a busy tone is audible to the other party indicating that the call connection breaks. 0-without a busy tone 3-3s 5-5s If it is set to 3 (3s), a busy tone is audible for 3 seconds on the IP phone.
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Configuring Basic Features return code received. Available return codes and reasons are:  404 (Not found)  480 (Temporarily not available)  486 (Busy here) Procedure Return code for call rejection can be configured using the configuration files or locally. Configure the return code when Configuration File y000000000037.cfg refusing a call. Parameter: features.normal_refuse_code Configure the return code when refusing a call.
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Administrator’s Guide for CP860 IP conference phones 2. Select the desired value from the pull-down list of Return Code When Refuse. 3. Click Confirm to accept the change. Early media refers to media (e.g., audio and video) played to the caller before a SIP call is actually established. Current implementation supports early media through the 183 message. When the caller receives a 183 message with SDP before the call is established, a media channel is established.
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Configuring Basic Features Details of the Configuration Parameter: Parameter phone_setting.is_deal180 Permitted Values Default 0 or 1 1 Description: Enables or disables the IP phone to deal with the 180 SIP message received after the 183 SIP message. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will resume and play the local ringback tone upon a subsequent 180 message received.
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Administrator’s Guide for CP860 IP conference phones An outbound proxy server can receive all initiating request messages and route them to the designated destination. If the IP phone is configured to use an outbound proxy server within a dialog, all SIP request messages from the IP phone will be sent to the outbound proxy server forcefully. Note To use this feature, make sure the outbound server has been correctly configured on the IP phone.
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Configuring Basic Features Parameter Permitted Values Default None To specify whether to use outbound proxy server in a dialog via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Use Outbound Proxy in Dialog. 3. Click Confirm to accept the change. SIP session timers T1, T2 and T4 are SIP transaction layer timers defined in RFC 3261.
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Administrator’s Guide for CP860 IP conference phones account.X.advanced.timer_t4 Configure SIP session timer. Navigate to: Local Web User Interface http:///servlet ?p=account-adv&q=load&acc= 0 Details of Configuration Parameters: Parameters account.X.advanced.timer_t1 (X = 1) Permitted Values Default Float from 0.5 to10 0.5 Description: Configures the SIP session timer T1 (in seconds).
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Configuring Basic Features Parameters Permitted Values Default Description: Configures the session timer of T4 (in seconds). T4 represents the maximum duration a message will remain in the network. Web User Interface: Account->Advanced->SIP Session Timer T4 (2.5~60s) Phone User Interface: None To configure session timer via web user interface: 1. Click on Account->Advanced. 2. Enter the desired value in the SIP Session Timer T1 (0.5~10s) field. The default value is 0.5s. 3.
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Administrator’s Guide for CP860 IP conference phones “c” (connection addresses for the media streams) in the SDP to zero (e.g., c=0.0.0.0). Call hold tone allows IP phones to play a warning tone at regular intervals when there is a call on hold. The warning tone is played through the speakerphone. IP phones also support Music on Hold (MoH) feature. MoH is the business practice of playing recorded music to fill the silence that would be heard by the party who has been placed on hold.
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Configuring Basic Features 0 Details of Configuration Parameters: Parameters Permitted Values Default 0 or 1 1 features.play_hold_tone.enable Description: Enables or disables the IP phone to play a tone when there is a call on hold. 0-Disabled 1-Enabled Web User Interface: Features->General Information->Play Hold Tone Phone User Interface: None features.play_hold_tone.delay Integer from 3 to 3600 30 Description: Configures the interval (in seconds) at which the IP phone plays a hold tone.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default SIP URI within 256 characters Blank Features->General Information->RFC 2543 Hold Phone User Interface: None account.X.music_server_uri (X = 1) Description: Configures the address of the Music On Hold server. Examples for valid values: <10.1.3.165>, 10.1.3.165, sip:moh@sip.com, , or yealink.com. Example: account.1.music_server_uri =<10.1.3.
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Configuring Basic Features To configure call hold tone and call hold tone delay via web user interface: 1. Click on Features->General Information. 2. Select the desired value from the pull-down list of Play Hold Tone. 3. Enter the desired time in the Play Hold Tone Delay field. 4. Click Confirm to accept the change. To configure MoH via web user interface: 1. Click on Account->Advanced.
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Administrator’s Guide for CP860 IP conference phones 2. Enter the SIP URI (e.g., sip:moh@sip.com) in the Music Server URI field. 3. Click Confirm to accept the change. Session timer allows a periodic refresh of SIP sessions through a re-INVITE request, to determine whether a SIP session is still active. Session timer is specified in RFC 4028. IP phones support two refresher modes: UAC and UAS.
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Configuring Basic Features 0 Details of Configuration Parameters: Parameters account.X.session_timer.enable (X = 1) Permitted Values Default 0 or 1 0 Description: Enables or disables the session timer. 0-Disabled 1-Enabled If it is set to 1 (Enabled), IP phone will send periodic re-INVITE requests to refresh the session during a call. Web User Interface: Account->Advanced->Session Timer Phone User Interface: None account.X.session_timer.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Description: Configures the session timer refresher. 0-UAC 1-UAS If it is set to 0 (UAC), refreshing the session is performed by the IP phone. If it is set to 1 (UAS), refreshing the session is performed by a SIP server. Web User Interface: Account-> Advanced-> Session Refresher Phone User Interface: None To configure session timer via web user interface: 138 1. Click on Account->Advanced. 2.
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Configuring Basic Features Call forward allows users to redirect an incoming call to a third party. IP phones redirect an incoming INVITE message by responding with a 302 Moved Temporarily message, which contains a Contact header with a new URI that should be tried. Three types of call forward:  Always Forward -- Forward the incoming calls immediately.  Busy Forward -- Forward the incoming call when the IP phone or the specified account is busy.
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Administrator’s Guide for CP860 IP conference phones forward.busy.off_code forward.no_answer.enable forward.no_answer.target forward.no_answer.timeout forward.no_answer.on_code forward.no_answer.off_code features.fwd_diversion_enable Configure forward international. Parameter: forward.international.enable Configure call forward. Navigate to: http:///servlet ?p=features-forward&q=load Web User Interface Local Configure forward international.
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Configuring Basic Features Parameters Permitted Values Default Description: Configures the destination number the IP phone forwards all incoming calls to. Web User Interface: Features->Forward &DND->Always Forward->Target Phone User Interface: Menu->Features->Call Forward->Always Forward->Forward To forward.always.on_code String within 32 characters Blank Description: Configures the always forward on code to activate the server-side always forward feature.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default 0-Disabled 1-Enabled If it is set to 1 (Enabled), incoming calls are forwarded to the destination number when the callee is busy. Web User Interface: Features->Forward &DND->Busy Forward->On/Off Phone User Interface: Menu->Features->Call Forward->Busy Forward->Busy Forward forward.busy.
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Configuring Basic Features Parameters Permitted Values Default Configures the busy forward off code to deactivate the server-side busy forward feature. The IP phone will send the busy forward off code to the server when you deactivate busy forward feature on the IP phone. Example： forward.busy.off_code = *76 Web User Interface: Features->Forward &DND->Busy Forward->Off Code Phone User Interface: Menu->Features->Call Forward->Busy Forward->Off Code forward.no_answer.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures ring times (N) to wait before forwarding incoming calls. Incoming calls will be forwarded when not answered after N*6 seconds. Web User Interface: Features->Forward &DND->No Answer Forward->After Ring Time (0~120s) Phone User Interface: Menu->Features->Call Forward->No Answer Forward->After Ring Time forward.no_answer.
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Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables the IP phone to present the diversion information when an incoming call is forwarded to your IP phone. 0-Disabled 1-Enabled Web User Interface: Features->General Information->Diversion/History-Info Phone User Interface: None forward.international.enable 0 or 1 1 Description: Enables or disables the IP phone to forward incoming calls to international numbers (the prefix is 00).
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Administrator’s Guide for CP860 IP conference phones 4) Select the ring time to wait before forwarding from the pull-down list of After Ring Time (0~120s) (only for the no answer forward). 3. Click Confirm to accept the change. To configure the forward international feature via web user interface: 146 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.
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Configuring Basic Features To enable call forward via phone user interface: 1. Press Menu->Features->Call Forward. 2. Press or to select the desired forwarding type, and then press the Enter soft key. 3. Depending on your selection: a.) If you select Always Forward: 1) Press the or soft key to select Enable from the Always Forward field. 2) Enter the destination number you want to forward all incoming calls to in the Forward to field. 3) (Optional.
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Administrator’s Guide for CP860 IP conference phones Semi-attended transfer is implemented by a REFER method with Replaces in the Refer-To header.  Attended Transfer -- Transfer a call with prior consulting. Attended transfer is implemented by a REFER method with Replaces in the Refer-To header. Normally, call transfer is completed by pressing the transfer key. Blind transfer on hook and attended transfer on hook features allow the IP phone to complete the transfer through pressing the on-hook key.
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Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables the IP phone to complete the blind transfer through pressing the on-hook key instead of pressing the Tran soft key. 0-Disabled 1-Enabled Web User Interface: Features->Transfer->Blind Transfer On Hook Phone User Interface: None transfer.
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Administrator’s Guide for CP860 IP conference phones 2. Select the desired values from the pull-down lists of Semi-Attended Transfer, Blind Transfer On Hook and Semi-Attend 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.
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Configuring Basic Features Parameters Permitted Values Default Description: Configures the network conference type. 0-Local Conference 2-Network Conference If it is set to 0 (Local Conference), conferences are set up on the IP phone locally. If it is set to 2 (Network Conference), conferences are set up by the server. Web User Interface: Account->Advanced->Conference Type Phone User Interface: None account.X.
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Administrator’s Guide for CP860 IP conference phones 3. Enter the conference URI in the Conference URI field. 4. Click Confirm to accept the change. For local conference, all parties drop the call when the conference initiator drops the conference call. For local conference, transfer on conference hang up allows the other two parties to remain connected when the conference initiator drops the conference call.
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Configuring Basic Features Details of the Configuration Parameter: Parameter & Description transfer.tran_others_after_conf_enable Permitted Values Default 0 or 1 0 Description: Enables or disables the IP phone to transfer the local conference call to the two parties after the conference initiator drops the local conference call. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the other two parties remain connected when the conference initiator drops the conference call.
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Administrator’s Guide for CP860 IP conference phones depends on support from a SIP server. For many SIP servers, directed call pickup requires a directed pickup code, which can be configured on a phone or per-line basis. Procedure Directed call pickup can be configured using the configuration files or locally. Configure the directed call .cfg pickup code on a per-line basis. Parameter: account.X.direct_pickup_code Configure directed call pickup Configuration File features on a phone basis.
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Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default 0 or 1 0 features.pickup.direct_pickup_enable Description: Enables or disables the IP phone to display the DPickup soft key when the IP phone is in the pre-dialing screen. 0-Disabled 1-Enabled Web User Interface: Features->Call Pickup->Directed Call Pickup Phone User Interface: None features.pickup.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Account->Advanced->Directed Call Pickup Code Phone User Interface: None 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.
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Configuring Basic Features 2. Enter the directed call pickup code in the Directed Call Pickup Code field. 3. Click Confirm to accept the change. Group call pickup is used for picking up incoming calls within a pre-defined group. If the group receives many incoming calls at once, the user will pick up the first incoming call by pressing the GPickup soft key. This feature depends on support from a SIP server.
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Administrator’s Guide for CP860 IP conference phones code Configure the group call pickup feature on a phone basis. Navigate to: http:///servl et?p=features-callpickup&q=lo ad Configure the group call pickup code on a phone basis. Local Web User Interface Navigate to: http:///servl et?p=features-callpickup&q=lo ad Configure the group call pickup code on a per-line basis.
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Configuring Basic Features Parameters Permitted Values Default Description： Configures the group call pickup code on a phone basis. Example: features.pickup.group_pickup_code = *98 Note: The group call pickup code configured on a per-line basis takes precedence over that configured on a phone basis. Web User Interface: Features->Call Pickup->Group Call Pickup Code Phone User Interface: None account.X.
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Administrator’s Guide for CP860 IP conference phones 3. Enter the group call pickup code in the Group Call Pickup Code field. 4. Click Confirm to accept the change. To configure the group call pickup code on a per-line basis via web user interface: 1. Click on Account->Advanced. 2. Enter the group call pickup code in the Group Call Pickup Code field. 3. Click Confirm to accept the change. Call return, also known as last call return, allows users to place a call back to the last caller.
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Configuring Basic Features Procedure Call return key can be configured using the configuration files or locally. Assign a call return key. Configuration File y000000000037.cfg Parameter: programablekey.X.type Assign a call return key. Local Web User Interface Navigate to: http:///servlet ?p=dsskey&model=2&q=load Details of Configuration Parameters: Parameter programablekey.X.
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Administrator’s Guide for CP860 IP conference phones 2. In the desired programable key field, select Call Return from the pull-down list of Type. 3. Click Confirm to accept the change. Calling line identification presentation (CLIP) allows IP phones to display the caller identity, derived from a SIP header contained in the INVITE message when receiving an incoming call. IP phones support deriving caller identity from three types of SIP header: From, P-Asserted-Identity and Remote-Party-ID.
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Configuring Basic Features Details of the Configuration Parameter: Parameter account.X.cid_source (X = 1) Permitted Values Default 0, 1, 2, 3, 4 or 5 0 Description: Configures the presentation of the caller identity when receiving an incoming call. 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).
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Administrator’s Guide for CP860 IP conference phones 3. Click Confirm to accept the change. Connected line identification presentation (COLP) allows IP phones to display the identity of the connected party specified for outgoing calls. IP phones can display the Dialed Digits, or the identity in a SIP header (Remote-Party-ID or P-Asserted-Identity) received, or the identity in the From header carried in the UPDATE message sent by the callee as described in RFC 4916.
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Configuring Basic Features Parameter Permitted Values Default From header. Web User Interface: None Phone User Interface: None 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 pressed on the IP phone generates one sinusoidal tone of two frequencies.
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Administrator’s Guide for CP860 IP conference phones configurable. IP phones default to 101 for the payload type, which use the definition to negotiate with the other end during call establishment. The RTP Event packet contains 4 bytes. The 4 bytes are distributed over several fields denoted as Event, End bit, R-bit, Volume and Duration. If the End bit is set to 1, the packet contains the end of the DTMF event. You can configure the sending times of the end RTP Event packet.
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Configuring Basic Features RTP Event packet. Navigate to: http:///servl et?p=features-general&q=loa d Details of Configuration Parameters: Parameters Permitted Values Default 0, 1, 2 or 3 1 account.X.dtmf.type (X = 1) Description: Configures the DTMF type. 0-INBAND 1-RFC 2833 2-SIP INFO 3-AUTO or SIP INFO If it is set to 0 (INBAND), DTMF digits are transmitted in the voice band. If it is set to 1 (RFC 2833), DTMF digits are transmitted by RTP Events compliant to RFC 2833.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default (X = 1) Description: Configures the DTMF info type when the DTMF type is configured as “SIP INFO”, “AUTO or SIP INFO”. 0-Disabled 1-DTMF-Relay 2-DTMF 3-Telephone-Event Web User Interface: Account->Advanced->DTMF Info Type Phone User Interface: None features.dtmf.repetition 1, 2 or 3 3 Description: Configures the repetition times for the IP phone to send the end RTP EVENT packet during an active call.
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Configuring Basic Features 3. Enter the desired value in the DTMF Payload Type (96~127) field. 4. Click Confirm to accept the change. To configure the number of times to send the end RTP Event packet via web user interface: 1. Click on Features->General Information. 2. Select the desired value (1-3) from the pull-down list of DTMF Repetition. 3. Click Confirm to accept the change. Suppress DTMF display allows IP phones to suppress the display of DTMF digits.
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Administrator’s Guide for CP860 IP conference phones whether to display the DTMF digits for a short period of time before displaying as “*”. Procedure Configuration changes can be performed using the configuration files or locally. Configure suppress DTMF display and suppress DTMF display delay. Configuration File y000000000037.cfg Parameters: features.dtmf.hide features.dtmf.hide_delay Configure suppress DTMF display and suppress DTMF display delay.
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Configuring Basic Features Parameters Permitted Values Default Description: Enables or disables the IP phone to display the DTMF digits for a short period before displaying asterisks during an active call. 0-Disabled 1-Enabled Note: It works only if the parameter “features.dtmf.hide” is set to 1 (Enabled). Web User Interface: Features->General Information->Suppress DTMF Display Delay Phone User Interface: None To configure suppress DTMF display and suppress DTMF display delay via web user interface: 1.
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Administrator’s Guide for CP860 IP conference phones Call transfer is implemented via DTMF on some traditional servers. The IP phone sends specified DTMF digits to the server for transferring calls to third parties. Procedure Configuration changes can be performed using the configuration files or locally. Configure transfer via DTMF. Configuration File Parameters: y000000000037.cfg features.dtmf.replace_tran features.dtmf.transfer Configure transfer via DTMF.
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Configuring Basic Features Parameters Permitted Values Default Description: Configures the DTMF digits to be transmitted to perform call transfer. Valid values are: 0-9, *, # and A-D. Example: features.dtmf.transfer = 123 Note: It works only if the parameter “features.dtmf.replace_tran” is set to 1 (Enabled). Web User Interface: Features->General Information->Tran Send DTMF Phone User Interface: None To configure transfer via DTMF feature via web user interface: 1.
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Administrator’s Guide for CP860 IP conference phones Intercom allows establishing an audio conversation directly. The IP phone can answer intercom calls automatically. This feature depends on support from a SIP server. Intercom is a useful feature in office environments to quickly connect with an operator or secretary. Users can press an intercom key to automatically initiate an outgoing intercom call with a remote extension.
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Configuring Basic Features Parameters programablekey.X.value (X=1-6, 9, 13) Permitted Values Default String within 99 characters blank Description: Configures the intercom number. Example: programablekey.2.value = 1008 Web User Interface: DSSKey->Programable Key->Value Phone User Interface: None To configure an intercom key via web user interface: 1. Click on DSSKey->Programable Key. 2. In the desired programable key field, select Intercom from the pull-down list of Type. 3.
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Administrator’s Guide for CP860 IP conference phones Intercom Mute Intercom Mute allows the IP phone to mute the microphone for incoming intercom calls. Intercom Tone Intercom Tone allows the IP phone to play a warning tone before answering an intercom call. Intercom Barge Intercom Barge allows the IP phone to automatically answer an incoming intercom call while an active call is in progress. The active call will be placed on hold.
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Configuring Basic Features Parameters Permitted Values Default 0 or 1 0 Web User Interface: Features->Intercom->Accept Intercom Phone User Interface: Menu->Features->Intercom->Accept Intercom features.intercom.mute Description: Enables or disables the IP phone to mute the microphone when answering an intercom call. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the microphone is muted for intercom calls, and then the other party cannot hear you. Note: It works only if the parameter “features.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Enables or disables the IP phone to automatically answer an incoming intercom call while there is already an active call on the IP phone. 0-Disabled 1-Enabled If it is set to 0 (Disabled), the IP phone will handle an incoming intercom call like a waiting call while there is already an active call on the IP phone.
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Configuring Advanced Features This chapter provides information for making configuration changes for the following advanced features:  Distinctive Ring Tones  Tones  Remote Phone Book  LDAP  Message Waiting Indicator  Multicast Paging  Action URL  Action URI  Server Redundancy  Static DNS Cache  LLDP  VLAN  VPN  Quality of Service  Network Address Translation  SNMP  802.
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Administrator’s Guide for CP860 IP conference phones Distinctive ring tones allows certain incoming calls to trigger IP phones to play distinctive ring tones. The IP phone inspects the INVITE request for an "Alert-Info" header when receiving an incoming call. If the INVITE request contains an "Alert-Info" header, the IP phone strips out the URL or keyword parameter and maps it to the appropriate ring tone.
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Configuring Advanced Features Bellcore Pattern Tone ID Minimum Nominal Maximum Duration Duration Duration (ms) (ms) (ms) 145 200 525 630 800 1025 2975 4000 4400 200 300 525 145 200 525 800 1000 1100 145 200 525 200 300 525 Silent 2975 4000 4400 Ringing 450 500 550 Pattern Cadence Silent Ringing Long Silent Ringing Short Silent Bellcore-dr4 Ringing 4 Silent Ringing Bellcore-dr5 Note Long 5 Short “Bellcore-dr5” is a ring splash tone that reminds the
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Administrator’s Guide for CP860 IP conference phones “Distinctive Ring Tones” on the web user interface is Enabled), or play the preconfigured local ring tone in about ten seconds if the parameter “account.X.alert_info_url_enable” is set to 0 or if the IP phone fails to download the remote ring tone. Example: Alert-Info: http://192.168.0.12:8080/Custom.
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Configuring Advanced Features http:///servlet?p=accou nt-adv&q=load&acc=0 Configure the internal ringer text and internal ringer file. Navigate to: http:///servlet?p=setting s-ring&q=load Details of Configuration Parameters: Parameters account.X.alert_info_url_enable (X = 1) Permitted Values Default 0 or 1 1 Description: Enables or disables the IP phone to download the ring tone from the URL contained in the Alert-Info header.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the internal ringer text to map the keywords contained in the Alert-Info header. Example: distinctive_ring_tones.alert_info.1.text = family Web User Interface: Settings->Ring->Internal Ringer Text Phone User Interface: None distinctive_ring_tones.alert_info.X.ringer (X ranges from 1 to 10) String within 32 characters Description: Configures the desired ring tones for each text.
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Configuring Advanced Features 2. Select the desired value from the pull-down list of Distinctive Ring Tones. 3. Click Confirm to accept the change. To configure the internal ringer text and internal ringer file via web user interface: 1. Click on Settings->Ring. 2. Enter the keywords in the Internal Ringer Text fields.
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Administrator’s Guide for CP860 IP conference phones 3. Select the desired ring tones for each text from the pull-down lists of Internal Ringer File. 4. Click Confirm to accept the change. When receiving a message, the IP phone will play a warning tone. You can customize tones or select specialized tone sets (vary from country to country) to indicate different conditions of the IP phone. The default tones used on IP phones are the US tone sets.
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Configuring Advanced Features  Great Britain  Greece  Hungary  Lithuania  India  Italy  Japan  Mexico  New Zealand  Netherlands  Norway  Portugal  Spain  Switzerland  Sweden  Russia  United States  Chile  Czech ETSI 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 congest
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Administrator’s Guide for CP860 IP conference phones Procedure Tones can be configured using the configuration files or locally. Configure the tones for the IP phone. Parameters: voice.tone.country voice.tone.dial voice.tone.ring Configuration File y000000000037.cfg voice.tone.busy voice.tone.congestion voice.tone.callwaiting voice.tone.dialrecall voice.tone.info voice.tone.stutter voice.tone.autoanswer Configure the tones for the IP phone.
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Configuring Advanced Features Parameters Permitted Values Default String Blank Phone User Interface: None voice.tone.dial Description: Customizes the dial tone. tonelist = element[,element] [,element]… Where element = [!]Freq1[+Freq2][+Freq3][+Freq4] /Duration Freq: the frequency of the tone (ranges from 200 to 7000 Hz). If it is set to 0Hz, it means the tone is not played. A tone is comprised of at most four different frequencies.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Customizes the tone when the callee is busy. The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”. The default value is blank. Note: It works only if the parameter “voice.tone.country” is set to Custom. Web User Interface: Settings->Tones->Busy Phone User Interface: None voice.tone.
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Configuring Advanced Features Parameters Permitted Values Default Description: Customizes the call back tone. The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”. Note: It works only if the parameter “voice.tone.country” is set to Custom. Web User Interface: Settings->Tones->Dial Recall Phone User Interface: None voice.tone.info String Blank Description: Customizes the info tone. The value format is Freq/Duration.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Customizes the warning tone for auto answer. The value format is Freq/Duration. For more information on the value format, refer to the parameter “voice.tone.dial”. The default value is blank. Note: It works only if the parameter “voice.tone.country” is set to Custom. Web User Interface: Settings->Tones->Auto Answer Phone User Interface: None To configure tones via web user interface: 1.
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Configuring Advanced Features remote phone book entries on the phone user interface. IP phones support up to 5 remote phone books and 5000 entries. Remote phone book is customizable. For more information, refer to Remote XML Phone Book on page 331. Sremote Name allows IP phones to search the entry names from the remote phone book for incoming/outgoing calls. Sremote Name Flash Time defines how often IP phones refresh the local cache of the remote phone book.
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Administrator’s Guide for CP860 IP conference phones Details of Configuration Parameters: Parameters remote_phonebook.data.X.url Permitted Values Default URL within 511 characters Blank (X ranges from 1 to 5) Description: Configures the access URL of the remote phone book. Example: remote_phonebook.data.1.url = http://192.168.1.20/Menu.xml Web User Interface: Directory->Remote Phone Book->Remote URL Phone User Interface: None remote_phonebook.data.X.
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Configuring Advanced Features Parameters features.remote_phonebook.flash_time Permitted Values Integer from 120 to 2592000 Default 21600 Description: Configures how often to refresh the local cache of the remote phone book. If it is set to 3600, the IP phone will refresh the local cache of the remote phone book every 3600 seconds.
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Administrator’s Guide for CP860 IP conference phones 3. Enter the desired time in the Search 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 for the distributed directory over an IP network. IP phones can be configured to interface with a corporate directory server that supports LDAP version 2 or 3.
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Configuring Advanced Features LDAP Attributes The following table lists the most common attributes used to configure the LDAP lookup on IP phones: Abbreviation Name gn givenName cn commonName sn surname dn distinguishedName dc dc - company - telephoneNumber mobile mobilephoneNumber ipPhone IPphoneNumber Description First name LDAP attribute is made up from given name joined to surname.
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Administrator’s Guide for CP860 IP conference phones ldap.call_in_lookup ldap.ldap_sort Assign an LDAP key. Parameter: programablekey.X.type Configure the LDAP feature. Navigate to: http:///servl et?p=contacts-LDAP&q=load Local Web User Interface Assign an LDAP key. Navigate to: http:///servl et?p=dsskey&model=2&q=loa d Details of Configuration Parameters: Parameters Permitted Values Default 0 or 1 0 String within 99 characters Blank ldap.
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Configuring Advanced Features Parameters Permitted Values Default String within 99 characters Blank Web User Interface: Directory->LDAP->LDAP Name Filter Phone User Interface: None ldap.number_filter Description: Configures the criteria for searching the LDAP contact number attributes. The “*” symbol in the filter stands for any character. The “%” symbol in the filter stands for the entering string used as the prefix of the filter condition. Example: ldap.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default String within 99 characters Blank Web User Interface: Directory->LDAP->Port Phone User Interface: None ldap.base Description: Configures the LDAP search base which corresponds to the location of the LDAP phone book from which the LDAP search request begins. The search base narrows the search scope and decreases directory search time. Example: ldap.
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Configuring Advanced Features Parameters Permitted Values Default Integer from 1 to 32000 50 ldap.password =secret Web User Interface: Directory->LDAP->Password Phone User Interface: None ldap.max_hits Description: Configures the maximum number of search results to be returned by the LDAP server. If the value of the “Max.Hits” is blank, the LDAP server will return all searched results. Please note that a very large value of the “Max.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the number attributes of each record to be returned by the LDAP server. It compresses the search results. You can configure multiple number attributes separated by spaces. Example: ldap.numb_attr = telephoneNumber Web User Interface: Directory->LDAP->LDAP Number Attributes Phone User Interface: None ldap.
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Configuring Advanced Features Parameters Permitted Values Default Description: Enables or disables the IP phone to perform an LDAP search when receiving an incoming call. 0-Disabled 1-Enabled Web User Interface: Directory->LDAP->LDAP Lookup For Incoming Call Phone User Interface: None ldap.ldap_sort 0 or 1 0 Description: Enables or disables the IP phone to sort the search results in alphabetical order or numerical order.
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Administrator’s Guide for CP860 IP conference phones To configure LDAP via web user interface: 1. Click on Directory->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->Programable Key. 2.
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Configuring Advanced Features when receiving new voice messages. IP phones support both solicited and unsolicited MWI. Unsolicited MWI is a server related feature. IP phone sends a SUBSCRIBE message to the server for message-summary updates. The server sends a message-summary NOTIFY within the subscription dialog each time the MWI status changes. For solicited MWI, you must enable MWI subscription feature on IP phones. IP phones support subscribing the MWI messages to the account or the voice mail number.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default 0-Disabled 1-Enabled Web User Interface: Account->Advanced->Subscribe for MWI Phone User Interface: None account.X.subscribe_mwi_expires (X = 1) Integer from 0 to 84600 3600 Description: Configures MWI subscribe expiry time (in seconds). The IP phone is able to successfully refresh the SUBSCRIBE for message-summary events before expiration of the SUBSCRIBE dialog. Note: It works only if the parameter “account.X.
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Configuring Advanced Features Parameters Permitted Values Default Configures the voice mail number. Example: voice_mail.number.1 = 1234 Note: It works only if the parameter “account.x.subscribe_mwi_to_vm” is set to 1 (Enabled). Web User Interface: Account->Advanced->Voice Mail Phone User Interface: None To configure subscribe for MWI via web user interface: 1. Click on Account->Advanced. 2. Select the desired value from the pull-down list of Subscribe for MWI. 3.
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Administrator’s Guide for CP860 IP conference phones 3. Enter the desired voice number in the Voice Mail field. 4. Click Confirm to accept the change. Multicast paging allows IP phones to send/receive Real-time Transport Protocol (RTP) streams to/from the pre-configured multicast address(es) without involving SIP signaling. Up to 10 listening multicast addresses can be specified on the IP phone. Users can send an RTP stream without involving SIP signaling by pressing a configured multicast paging key.
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Configuring Advanced Features Parameters: programablekey.X.type programablekey.X.value. Assign a multicast paging key. Navigate to: http:///servlet ?p=dsskey&model=2&q=load Local Web User Interface Specify a multicast codec for the IP phone to send the RTP stream. Navigate to: http:///servlet ?p=features-general&q=load Details of the Configuration Parameter: Parameters multicast.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Web User Interface: DSSKey->Programable Key->Programable KeyX->Type Phone User Interface: None programablekey.X.value String within 99 (X=1-6, 9, 13) characters blank Description: Configures the multicast IP address and port number. Example： programablekey.3.value = 224.5.5.6:10008 Note: The valid multicast IP addresses range from 224.0.0.0 to 239.255.255.255.
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Configuring Advanced Features 2. Select the desired codec from the pull-down list of Multicast Codec. 3. Click Confirm to accept the change. IP phones can receive an RTP stream from the pre-configured multicast address(es) without involving SIP signaling, and can handle the incoming multicast paging calls differently depending on the configurations of Paging Barge and Paging Priority Active.
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Administrator’s Guide for CP860 IP conference phones Procedure Configuration changes can be performed using the configuration files or locally. Configure the listening multicast address. Parameters: multicast.listen_address.X.label Configuration File multicast.listen_address.X.ip_address y000000000037.cfg Configure the Paging Barge and Paging Priority Active features. Parameters: multicast.receive_priority.enable multicast.receive_priority.priority Configure the listening multicast address.
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Configuring Advanced Features Parameters Permitted Values Default Description: Configures the label to be displayed on the LCD screen when receiving the RTP multicast. Example: multicast.listen_address.1.label = Paging1 Web User Interface: Directory->Multicast IP->Label Phone User Interface: None multicast.receive_priority.enable 0 or 1 1 Description: Enables or disables the IP phone to handle the incoming multicast paging calls when there is an active multicast paging call on the IP phone.
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Administrator’s Guide for CP860 IP conference phones 2. Enter the listening multicast address and port number in the Listening Address field. 1 is the highest priority and 10 is the lowest priority. 3. Enter the label in the Label field. The label will appear on the LCD screen when receiving the RTP multicast. 4. Click Confirm to accept the change. To configure the paging barge and paging priority active features via web user interface: 214 1. Click on Directory->Multicast IP. 2.
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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 only be triggered by the pre-defined events (e.g., log on). The valid URL format is: http(s)://IP address of the server/help.xml?. The following table lists the pre-defined events for action URL. Event Description Setup Completed When the IP phone completes startup.
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Administrator’s Guide for CP860 IP conference phones Event Description UnMute When the IP phone un-mutes a call. Missed Call When the IP phone misses a call. IP Changed When the IP address of the 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. Transfer Finished When the IP phone completes to transfer a call.
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Configuring Advanced Features Variable Value Description call. The SIP URI of the callee when the IP phone receives an incoming call. The SIP URI of the callee when the IP phone places a $remote call. The SIP URI of the caller when the IP phone receives an incoming call. The display name of the caller when the IP phone $display_local places a call. The display name of the callee when the IP phone receives an incoming call.
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Administrator’s Guide for CP860 IP conference phones action_url.no_answer_fwd_on action_url.no_answer_fwd_off action_url.transfer_call action_url.blind_transfer_call action_url.attended_transfer_call action_url.hold action_url.unhold action_url.mute action_url.unmute action_url.missed_call action_url.call_terminated action_url.busy_to_idle action_url.idle_to_busy action_url.ip_change action_url.forward_incoming_call action_url.reject_incoming_call action_url.answer_new_incoming_call action_url.
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Configuring Advanced Features Parameters  $active_url  $active_user  $active_host  $local  $remote  $display_local  $display_remote  $call_id Permitted Values Default Example: action_url. setup_completed = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Setup Completed action_url.registered URL within 511 characters Blank Description: Configures the action URL the IP phone sends after an account is registered. Example: action_url.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the action URL the IP phone sends when a register failed. Example: action_url.register_failed = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Register Failed Phone User Interface: None action_url.off_hook URL within 511 characters Blank Description: Configures the action URL the IP phone sends when off hook. Example: action_url.off_hook = http://192.168.0.
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Configuring Advanced Features Parameters Permitted Values Default action_url.incoming_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Incoming Call Phone User Interface: None action_url.outgoing_call URL within 511 characters Blank Description: Configures the action URL the IP phone sends when placing a call. Example: action_url.outgoing_call = http://192.168.0.20/help.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Phone User Interface: None action_url.dnd_off URL within 511 characters Blank Description: Configures the action URL the IP phone sends when DND feature is disabled. Example: action_url.dnd_off = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Close DND Phone User Interface: None action_url.
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Configuring Advanced Features Parameters action_url.busy_fwd_on Permitted Values URL within 511 characters Default Blank Description: Configures the action URL the IP phone sends when busy forward feature is enabled. Example: action_url.busy_fwd_on = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Open Busy Forward Phone User Interface: None action_url.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the action URL the IP phone sends when no answer forward feature is disabled. Example: action_url.no_answer_fwd_off = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Close No Answer Forward Phone User Interface: None action_url.transfer_call URL within 511 characters Blank Description: Configures the action URL the IP phone sends when performing a transfer.
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Configuring Advanced Features Parameters Permitted Values Default Example: action_url.attended_transfer_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Attended Transfer Phone User Interface: None action_url.hold URL within 511 characters Blank Description: Configures the action URL the IP phone sends when placing a call on hold. Example: action_url.hold = http://192.168.0.20/help.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Features->Action URL->Mute Phone User Interface: None action_url.unmute URL within 511 characters Blank Description: Configures the action URL the IP phone sends when un-muting a call. Example: action_url.unmute = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->UnMute Phone User Interface: None action_url.
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Configuring Advanced Features Parameters action_url.busy_to_idle Permitted Values URL within 511 characters Default Blank Description: Configures the action URL the IP phone sends when changing the state of the IP phone from busy to idle. Example: action_url.busy_to_idle = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Busy To Idle Phone User Interface: None action_url.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the action URL the IP phone sends when forwarding an incoming call. Example: action_url.forward_incoming_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Forward Incoming Call Phone User Interface: None action_url.
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Configuring Advanced Features Parameters Permitted Values Default action_url.transfer_finished = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Transfer Finished Phone User Interface: None action_url.transfer_failed URL within 511 characters Blank Description: Configures the action URL the IP phone sends when failing to transfer a call. Example: action_url.transfer_failed = http://192.168.0.20/help.
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Administrator’s Guide for CP860 IP conference phones 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 “=”. The valid URI format is: http(s)://phone IP address/servlet?key=variable value.
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Configuring Advanced Features Variable Value Note Phone Action BTrans=xxx Perform a blind transfer to xxx. CALLEND End a call. The variable value is not applicable to all events. For example, the variable value “MUTE” is only applicable when the IP phone is during a call. When authentication is required, you must enter “p=login&q=login&username=xxx&pwd=yyy&jumpto=URI&” before the variable “key”. xxx refers to the login user name, and yyy refers to the login password.
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Administrator’s Guide for CP860 IP conference phones Parameter Permitted Values Default 0~255. For example: 10.10.*.* stands for the IP addresses that range from 10.10.0.0 to 10.10.255.255. If it is left blank, the IP phone cannot receive or handle any HTTP GET request. If it is set to “any”, the IP phone will accept and handle HTTP GET requests from any IP address. Example: features.
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Configuring Advanced Features Two types of redundancy are possible. In some cases, a combination of the two may be deployed:  Failover: In this mode, the full phone system functionality is preserved by having a second equivalent capability call server take over from the one that has gone down/off-line. This mode of operation should be done using the DNS mechanisms from the primary to the secondary server.
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Administrator’s Guide for CP860 IP conference phones Phone Registration Two registration methods for fallback mode:  Concurrent registration: The IP phone registers to two SIP servers (working server and fallback server) at the same time. In a failure situation, a fallback server can take over the basic calling capability, but without some of the advanced features offered by the working server (default registration method).  Successive registration: The IP phone only registers to one server at a time.
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Configuring Advanced Features Details of Configuration Parameters: Parameters Permitted Values account.X.sip_server.Y.address String within 256 characters (X = 1, Y ranges from 1 to 2) Default Blank Description: Configures the IP address or domain name of the SIP server Y. Example: account.1.sip_server.1.address = yealink.pbx.com Web User Interface: Account->Register->SIP Server Y->Server Host Phone User Interface: None account.X.sip_server.Y.
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Administrator’s Guide for CP860 IP conference phones Parameters account.X.sip_server.Y.retry_counts (X = 1, Y ranges from 1 to 2) Permitted Values Default Integer from 0 to 20 3 Description: Configures the retry times for the IP phone to resend requests when the SIP server Y is unavailable or there is no response from the SIP server Y. Web User Interface: Account->Register->SIP Server Y->Server Retry Counts Phone User Interface: None account.X.fallback.
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Configuring Advanced Features Parameters Permitted Values Default Description: Configures the way in which the phone fails back to the primary server for call control in the failover mode. 0-newRequests: all requests are sent to the primary server first, regardless of the last server that was used. 1-DNSTTL: the IP phone will send requests to the last registered server first. If the time defined by DNSTTL on the registered server expires, the phone will retry to send requests to the primary server.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default 1-Enabled Web User Interface: None Phone User Interface: None To configure server redundancy for fallback purpose via web user interface: 1. Click on Account->Register. 2. Configure registration parameters of the account in the corresponding fields. 3. Select the desired value from the pull-down list of Transport. 4. Configure parameters of SIP server 1 and SIP server 2 in the corresponding fields. 5.
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Configuring Advanced Features 4. Configure parameters of the SIP server 1 or SIP server 2 in the corresponding fields. You must set the port of SIP server to 0 for NAPTR, SRV and A queries. 5. Note Click Confirm to accept the change. If the outbound proxy server is required and the transport is set to DNS-NAPTR, you must set the port of outbound proxy server to 0 for NAPTR, SRV and A queries.
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Administrator’s Guide for CP860 IP conference phones NAPTR (Naming Authority Pointer) First, the IP phone sends the NAPTR query to get the SRV pointer and transport protocol. Example of NAPTR records: order pref flags service regexp replacement IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.yealink.pbx.com IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.yealink.pbx.
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Configuring Advanced Features Parameters are explained in the following table: Parameter Priority Description Specify preferential treatment for the specific host entry. Lower priority is MORE preferred. When priorities are equal, weight is used to differentiate the Weight preference. The preference is from highest to lowest. Keep the same to load balance. Port Target Identify the port number to be used. Identify the actual host for an A query. SRV query returns two records.
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Administrator’s Guide for CP860 IP conference phones If it is not the last server in the list, the maximum number of retries depends on the configured retry count. Procedure SIP Server Domain Name Resolution can be configured using the configuration files or locally. Configure the transport type on the IP phone. Configuration File .cfg Parameters: account.X.transport account.X.naptr_build Configure the transport type on the IP phone.
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Configuring Advanced Features Parameters Permitted Values Default Description: Configures the way of SRV query for the IP phone to be performed when no result is returned from NAPTR query. 0-SRV query using UDP only 1-SRV query using UDP, TCP and TLS. Web User Interface: None Phone User Interface: None Failover redundancy can only be utilized when the configured domain name of the SIP server is resolved to multiple IP addresses.
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Administrator’s Guide for CP860 IP conference phones Procedure Static DNS cache can be configured only using the configuration files. Configure NAPTR/SRV/A records. Parameters: account.X.dns_cache_naptr.Y.name account.X.dns_cache_naptr.Y.flags account.X.dns_cache_naptr.Y.order account.X.dns_cache_naptr.Y.preference account.X.dns_cache_naptr.Y.replace account.X.dns_cache_naptr.Y.service account.X.dns_cache_naptr.Y.ttl account.X.dns_cache_srv.Y.name account.X.dns_cache_srv.Y.port account.X.dns_cache_srv.Y.
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Configuring Advanced Features Parameters Permitted Values Default account.1.dns_cache_naptr.1.name = yealink.pbx.com Web User Interface: None Phone User Interface: None account.X.dns_cache_naptr.Y.flags S, A, U or P (X= 1, Y ranges from 1 to 12) Blank Description: Configures the flag of NAPTR record Y. (Always “s” for SIP, which means to do an SRV lookup on whatever is in the replacement field). S-Do an SRV lookup next. A-Do an A lookup next. U-No need to do a DNS query next.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default (X= 1, Y ranges from 1 to 12) Description: Configures the preference of NAPTR record Y. NAPTR record with lower preference is more preferred. Example: account.X.dns_cache_naptr.Y.preference = 50 Web User Interface: None Phone User Interface: None account.X.dns_cache_naptr.Y.
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Configuring Advanced Features Parameters Permitted Values Default Phone User Interface: None account.X.dns_cache_naptr.Y.ttl Integer from 30 to 2147483647 (X= 1, Y ranges from 1 to 12) 300 Description: Configures the time interval (in seconds) that NAPTR record Y may be cached before the record should be consulted again. Example: account.1.dns_cache_naptr.1.ttl = 300 Web User Interface: None Phone User Interface: None account.X.dns_cache_srv.Y.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Integer from 0 to 65535 0 Phone User Interface: None account.X.dns_cache_srv.Y.priority (X= 1, Y ranges from 1 to 12) Description: Configures the priority for the target host in SRV record Y. Lower priority is more preferred. Web User Interface: None Phone User Interface: None account.X.dns_cache_srv.Y.
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Configuring Advanced Features Parameters Permitted Values Default None account.X.dns_cache_srv.Y.ttl Integer from 30 to (X= 1, Y ranges from 1 to 12) 2147483647 300 Description: Configures the time interval (in seconds) that SRV record Y may be cached before the record should be consulted again. Example: account.1.dns_cache_srv.1.ttl = 3600 Web User Interface: None Phone User Interface: None account.X.dns_cache_a.Y.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values account.X.dns_cache_a.Y.ttl Integer from 30 to (X= 1, Y ranges from 1 to 12) 2147483647 Default 300 Description: Configures the time interval (in seconds) that A record Y may be cached before the record should be consulted again. Example: account.1.dns_cache_a.1.ttl = 300 Web User Interface: None Phone User Interface: None account.X.
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Configuring Advanced Features Parameters Permitted Values Default account.1.static_cache_pri = 1 Web User Interface: None Phone User Interface: None LLDP (Linker Layer Discovery Protocol) is a vendor-neutral Link Layer protocol, which allows IP phones to receive and/or transmit device-related information from/to directly connected devices on the network that are also using the protocol, and store the information about other devices.
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Administrator’s Guide for CP860 IP conference phones TLV Type TLV Name End of LLDPDU System Name System Description Description Marks end of LLDPDU. Name assigned to the IP phone. The default value is “yealink”. Description of the IP phone. The default value is “yealink”. The supported and enabled phone capabilities. Optional TLVs System Capabilities The supported capabilities are Bridge, Telephone and Router. The enabled capabilities are Bridge and Telephone by default.
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Configuring Advanced Features TLV Type TLV Name Inventory – Firmware Revision Inventory – Software Revision Inventory – Serial Number Description Firmware revision of phone. Software revision of phone. Serial number of phone. Inventory – Manufacturer name of phone. Manufacturer Name The default value is “yealink”. Inventory – Model Name Asset ID Model name of phone. Assertion identifier of phone. The default value is “asset”.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Integer from 1 to 3600 60 Phone User Interface: None network.lldp.packet_interval Description: Configures the interval (in seconds) for the IP phone to broadcast the LLDP request. 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).
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Configuring Advanced Features 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. Grouping devices with a common set of requirements regardless of their physical location can greatly simplify network design. VLANs can address issues such as scalability, security, and network management.
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Administrator’s Guide for CP860 IP conference phones feature. Navigate to: http:///servlet? p=network-adv&q=load Configure VLAN for the Internet Phone User Interface port. Details of Configuration Parameters: Parameters Permitted Values Default 0 or 1 0 network.vlan.internet_port_enable Description: Enables or disables VLAN for the Internet (WAN) port. 0-Disabled 1-Enabled Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Configuring Advanced Features Parameters Permitted Values Default Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Advanced->VLAN ->WAN Port->Priority Phone User Interface: Menu->Settings->Advanced Settings (Default password: admin) ->Network-> VLAN ->WAN Port-> Priority network.vlan.dhcp_enable 0 or 1 1 Description: Enables or disables DHCP VLAN discovery feature on the IP phone.
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Administrator’s Guide for CP860 IP conference phones 4. Select the desired value (0-7) from the pull-down list of Priority. 5. Click Confirm to accept the change. A dialog box pops up to prompt reboot to make the settings effective. 6. Click OK to reboot the phone. 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.
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Configuring Advanced Features The default option is 132. 4. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after reboot. 5. Click OK to reboot the phone. To configure VLAN for Internet port via phone user interface: 1. Press Menu->Settings->Advanced Settings (Default password: admin) ->Network->VLAN->WAN Port. 2. Press the or soft key to select the desired value from the VLAN Status field. 3. Enter the VLAN ID (1-4094) in the VID field. 4.
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Administrator’s Guide for CP860 IP conference phones 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. VPN can be also classified by the protocols used to tunnel the traffic. It provides security through tunneling protocols: IPSec, SSL, L2TP and PPTP. IP phones support SSL VPN, which provides remote-access VPN capabilities through SSL.
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Configuring Advanced Features Details of Configuration Parameters: Parameters Permitted Values Default 0 or 1 0 network.vpn_enable Description： Enables or disables OpenVPN feature on the IP phone. 0-Disabled 1-Enabled Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Advanced->VPN->Active Phone User Interface: Menu->Settings->Advanced Settings (Default password: admin) ->Network->VPN->VPN Active openvpn.
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Administrator’s Guide for CP860 IP conference phones 3. Click Upload to upload the TAR file. The web user interface prompts the message “Import config…”. 4. In the VPN block, select the desired value from the pull-down list of Active. 5. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after reboot. 6. Click OK to reboot the phone. To configure VPN via phone user interface after uploading the tar file: 1.
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Configuring Advanced Features be considered when configuring a modern QoS implementation: bandwidth, delay, jitter and loss. QoS provides better network service through 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.
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Administrator’s Guide for CP860 IP conference phones Voice QoS In order to make VoIP transmissions intelligible to receivers, voice packets should not be dropped, excessively delayed, made to suffer varying delay. DiffServ model can guarantee high-quality voice transmission when the voice packets are configured to a higher DSCP value. SIP QoS SIP protocol is used for creating, modifying and terminating two-party or multi-party sessions.
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Configuring Advanced Features Parameters Permitted Values Default Integer from 0 to 63 26 None network.qos.signaltos Description： Configures the DSCP for SIP packets. The default DSCP value for SIP packets is 26 (Assured Forwarding). Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Advanced->SIP QoS (0~63) Phone User Interface: None To configure DSCPs for voice packets and SIP packets via web user interface: 1.
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Administrator’s Guide for CP860 IP conference phones Network Address Translation (NAT) is essentially a translation table that maps public IP address and port combinations to private ones. This reduces the need for a large number of public IP addresses. The NAT feature ensures security since each outgoing or incoming request must first go through a translation process. But in the VoIP environment, NAT breaks end-to-end connectivity.
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Configuring Advanced Features Details of Configuration Parameters: Parameters Permitted Values Default 0 or 1 0 IP address or domain name Blank account.X.nat.nat_traversal (X = 1) Description： Enables or disables the NAT traversal. 0-Disabled 1-Enabled Web User Interface: Account->Register->NAT Phone User Interface: None account.X.nat.stun_server (X = 1) Description： Configures the IP address or the domain name of the STUN server. Example: account.1.nat.stun_server = 218.107.220.
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Administrator’s Guide for CP860 IP conference phones To configure the NAT traversal and STUN server via web user interface: 1. Click on Account. 2. Select STUN from the pull-down list of NAT. 3. Enter the IP address or the domain name in the STUN Server field. 4. Click Confirm to accept the change. SNMP (Simple Network Management Protocol) is an Internet-standard protocol for managing devices on IP networks.
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Configuring Advanced Features MIB OID Description person for the IP phone, together with the contact information. For example, Sysadmin (root@localhost) An administratively-assigned name for YEALINK-MIB 1.3.6.1.2.1.37459.2.1.2.0 the IP phone. If the name is unknown, the value is a zero-length string. YEALINK-MIB 1.3.6.1.2.1.37459.2.1.3.0 The physical location of the IP phone. The time (in milliseconds) since the YEALINK-MIB 1.3.6.1.2.1.37459.2.1.4.
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Administrator’s Guide for CP860 IP conference phones Configure SNMP. Local Navigate to: Web User Interface http:///servl et?p=network-adv&q=load Details of Configuration Parameters: Parameters Permitted Values Default 0 or 1 0 network.snmp.enable Description: Enables or disables SNMP feature on the IP phone. 0-Disabled 1-Enabled Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Configuring Advanced Features Parameters Permitted Values Default If it is set to “0.0.0.0”, the IP phone can accept and handle GET requests from any IP address. If it is left blank, the IP phone cannot receive or handle any GET request. Example: network.snmp.trust_ip = 192.168.1.50 as.manager.com Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Administrator’s Guide for CP860 IP conference phones 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. IEEE 802.1X authentication is an IEEE standard for Port-based Network Access Control (PNAC), part of the IEEE 802.1 group of networking protocols. It offers an authentication mechanism for devices to connect to a LAN or WLAN. The 802.
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Configuring Advanced Features Details of Configuration Parameters: Parameters network.802_1x.mode Permitted Values Default 0, 1, 2, 3 or 4 0 Description: Configures the 802.1x authentication method. 0-Disabled 1-EAP-MD5 2-EAP-TLS 3-PEAP-MSCHAPv2 4-EAP-TTLS/EAP-MSCHAPv2 Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Network->Advanced->802.1x->802.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Note: If you change this parameter, the IP phone will reboot to make the change take effect. It is required for all 802.1x authentication methods except EAP-TLS. Web User Interface: Network->Advanced->802.1x->MD5 Password Phone User Interface: Menu->Settings->Advanced Settings (Default password: admin) ->Network->802.1x Settings->MD5 Password network.802_1x.
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Configuring Advanced Features To configure the 802.1X via web user interface: 1. Click on Network->Advanced. 2. In the 802.1x block, select the desired protocol from the pull-down list of 802.1x Mode. a) b) If you select EAP-MD5: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. If you select EAP-TLS: 1) Enter the user name for authentication in the Identity field. 2) Leave the MD5 Password field blank.
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Administrator’s Guide for CP860 IP conference phones 5) c) Click Upload to upload the certificates. If you select PEAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to locate the desired certificate (*.pem,*.crt, *.cer or *.der) from your local system. 4) 276 Click Upload to upload the certificate.
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Configuring Advanced Features d) If you select EAP-TTLS/EAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. 3) In the CA Certificates field, click Browse to locate the desired certificate (*.pem,*.crt, *.cer or *.der) from your local system. 4) 3. Click Upload to upload the certificate. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot.
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Administrator’s Guide for CP860 IP conference phones d) 3. If you select EAP-TTLS/EAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field. Click Save to accept the change. The IP phone reboots automatically to make the settings effective after a period of time.
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Configuring Advanced Features RPC Method Description This method is used to cause the CPE to download a specified file from the designated location. File types supported by IP phones are: Download  Firmware Image  Configuration File This method is used to cause the CPE to upload a specified file to the designated location.
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Administrator’s Guide for CP860 IP conference phones managementserver.periodic_inform_interval Configure the TR-069 feature. Local Web User Navigate to: Interface http:///servlet?p=settings-prefer ence&q=load Details of Configuration Parameters: Parameters managementserver.enable Permitted Values Default 0 or 1 0 Description: Enables or disables TR-069 feature. 0-Disabled 1-Enabled Web User Interface: Settings->TR069->Enable TR069 Phone User Interface: None managementserver.
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Configuring Advanced Features Parameters Permitted Values Default required. Example: managementserver.password = pwd123 Web User Interface: Settings->TR069->ACS Password Phone User Interface: None managementserver.url URL within 511 characters Blank Description: Configures the access URL of the ACS (Auto Configuration Servers). Example: managementserver.url = http://192.168.1.20/acs/ Web User Interface: Settings->TR069->ACS URL Phone User Interface: None managementserver.
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Administrator’s Guide for CP860 IP conference phones Permitted Parameters Values Default Configures the password for the IP phone to authenticate the incoming connection requests. Example: managementserver.connection_request_password = acspwd Web User Interface: Settings->TR069->Connection Request Password Phone User Interface: None managementserver.
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Configuring Advanced Features 4. Enter the URL of the ACS in the ACS URL field. 5. Select the desired value from the pull-down list of Enable Periodic Inform. 6. Enter the desired time in the Periodic Inform Interval (seconds) field. 7. Enter the user name and password authenticated by the IP phone in the Connection Request Username and Connection Request Password fields. 8. Click Confirm to accept the change.
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Administrator’s Guide for CP860 IP conference phones Procedure IPv6 can be configured using the configuration files or locally. Configure the IPv6 address assignment method. Parameters： network.ip_address_mode network.ipv6_internet_port.type Configuration File network.ipv6_internet_port.ip .cfg network.ipv6_prefix network.ipv6_internet_port.gateway network.ipv6_primary_dns network.ipv6_secondary_dns network.ipv6_static_dns_enable Configure the IPv6 address assignment method.
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Configuring Advanced Features Parameters Permitted Values Default 0 or 1 0 Port->IP Mode network.ipv6_internet_port.type Description: Configures the Internet (WAN) port type for IPv6 when the IP address mode is configured as IPv6 or IPv4&IPv6. 0-DHCP 1-Static IP Address Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default effect. Web User Interface: Network->Basic->IPv6 Config->Static IP Address->IP Address Phone User Interface: Menu->Settings->Advanced Settings (Default password: admin)->Network->WAN Port->IPv6->Static IPv6 Client->IPv6 Address network.
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Configuring Advanced Features Parameters Permitted Values Default Description: Configures the primary IPv6 DNS server when the IP address mode is configured as IPv6 or IPv4&IPv6, and the Internet (WAN) port type for IPv6 is configured as Static IP Address. Example: network.ipv6_primary_dns = 3036:1:1:c3c7: c11c:5447:23a6:256 Note: If you change this parameter, the IP phone will reboot to make the change take effect.
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Administrator’s Guide for CP860 IP conference phones 3. In the IPv6 Config block, mark the DHCP or the Static IP Address radio box. - If you mark the Static IP Address radio box, configure the IPv6 address and other configuration parameters in the corresponding fields. - (Optional.) If you mark the DHCP radio box, you can configure the static DNS address in the corresponding fields. 4. 288 Click Confirm to accept the change.
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Configuring Advanced Features A dialog box pops up to prompt that the settings will take effect after reboot. 5. Click OK to reboot the phone. To configure IPv6 address via phone user interface: 1. Press Menu->Settings->Advanced Settings (Default password: admin) ->Network->WAN Port. 2. Press the or soft key to select the desired address mode from the IP Mode field. 3. Press to highlight IPv6 and press the Enter soft key. 4. Press 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:  Audio Codecs  Acoustic Clarity Technology CODEC is an abbreviation of COmpress-DECompress, capable of coding or decoding a digital data stream or signal by implementing an algorithm. The object of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining the quality.
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Administrator’s Guide for CP860 IP conference phones Codec Algorithm Reference Bit Rate Sample Packetization Rate Time 15.2 Kbps Packetization Time Ptime (Packetization Time) is a measurement of the duration (in milliseconds) of the audio data in each RTP packet sent to the destination, and defines how much network bandwidth is used for the RTP stream transfer. Before establishing a conversation, codec and ptime are negotiated through SIP signaling.
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Configuring Audio Features Codec Configuration Methods Configuration Files iLBC Web User Interface Priority RTPmap 4 106 Procedure Configuration changes can be performed using the configuration files or locally. Configure the codecs to use. Parameters: account.X.codec.Y.enable account.X.codec.Y.payload_type Configure the priority and rtpmap for the enabled codec. Parameters: Configuration File account.X.codec.Y.priority .cfg account.X.codec.Y.rtpmap Configure the display name of the codec.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Enables or disables the specified codec.
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Configuring Audio Features Parameters Permitted Values Default When Y=9, the default value is G726-24; When Y=10, the default value is G726-32; When Y=11, the default value is G726-40. Example: account.1.codec.1.payload_type = PCMU Web User Interface: Account->Codec Phone User Interface: None account.X.codec.Y.priority (X = 1, Y ranges from 1 to 11) Integer from 0 to 11 Refer to the following content Description: Configures the priority of the enabled codec.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default When Y=2, the default value is 8; When Y=3, the default value is 4; When Y=4, the default value is 4; When Y=5, the default value is 18; When Y=6, the default value is 9; When Y=7, the default value is 106; When Y=8, the default value is 103; When Y=9, the default value is 104; When Y=10, the default value is 102; When Y=11, the default value is 105; Web User Interface: None Phone User Interface: None account.X.
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Configuring Audio Features To configure the codecs and adjust the priority of the enabled codecs on a per-line basis via web user interface: 1. Click on Account->Codec. 2. Select the desired codec from the Disable Codecs column and click . The selected codec appears in the Enable Codecs column. 3. Repeat the step 2 to add more codecs to the Enable Codecs column. 4. Click to remove the codec from the Enable Codecs column. 5. Click or 6. Click Confirm to accept the change.
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Administrator’s Guide for CP860 IP conference phones Acoustic Echo Cancellation (AEC) is used to reduce acoustic echo from a voice call to provide natural full-duplex communication patterns. It also increases the capacity achieved through silence suppression by preventing echo from traveling across a network. Procedure AEC can be configured using the configuration files or locally. Configure AEC. Configuration File y000000000037.cfg Parameter: voice.echo_cancellation Configure AEC.
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Configuring Audio Features 2. Select the desired value from the pull-down list of ECHO. 3. Click Confirm to accept the change. Background noise suppression (BNS) is designed primarily for hands-free operation and reduces background noise to enhance communication in noisy environments. Automatic Gain Control (AGC) is applicable to hands-free operation and is used to keep audio output at nearly a constant level by adjusting the gain of signals in certain circumstances.
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Administrator’s Guide for CP860 IP conference phones voice.vad Configure VAD. Local Web User Interface Navigate to: http:///servl et?p=settings-voice&q=load Details of the Configuration Parameter: Parameter Permitted Values Default 0 or 1 0 voice.vad Description: Enables or disables VAD (Voice Activity Detection) feature on the IP phone.
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Configuring Audio Features Comfort Noise Generation (CNG) is used to generate background noise for voice communications during periods of silence in a conversation. It is a part of the silence suppression or VAD handling for VoIP technology. CNG, in conjunction with VAD algorithms, quickly responds when periods of silence occur and inserts artificial noise until voice activity resumes.
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Administrator’s Guide for CP860 IP conference phones 2. Select the desired value from the pull-down list of CNG. 3. Click Confirm to accept the change. Jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in even intervals. Jitter is a term indicating variations in packet arrival time, which can occur because of network congestion, timing drift or route changes.
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Configuring Audio Features jitter buffer. Navigate to: http:///servl et?p=settings-voice&q=load Details of Configuration Parameters: Parameters Permitted Values Default 0 or 1 1 Integer from 0 to 400 60 voice.jib.adaptive Description: Configures the type of jitter buffer. 0-Fixed 1-Adaptive Web User Interface: Settings->Voice->JITTER BUFFER->Type Phone User Interface: None voice.jib.min Description: Configures the minimum delay time (in milliseconds) of jitter buffer.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the normal delay time (in milliseconds) of jitter buffer. Note: It works only if the parameter “voice.jib.adaptive” is set to 0 (Fixed). Web User Interface: Settings->Voice->JITTER BUFFER->Normal Phone User Interface: None To configure Jitter Buffer via web user interface: 1. Click on Settings->Voice. 2. Mark the desired radio box in the Type field. 3.
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Configuring Security Features This chapter provides information for making configuration changes for the following security-related features:  Transport Layer Security  Secure Real-Time Transport Protocol  Encrypting Configuration Files TLS is a commonly-used protocol for providing communications privacy and managing the security of message transmission, allowing IP phones to communicate with other remote parties and connect to the HTTPS URL for provisioning in a way that is designed to prevent ea
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Administrator’s Guide for CP860 IP conference phones  AES256-SHA  EDH-RSA-DES-CBC3-SHA  EDH-DSS-DES-CBC3-SHA  DES-CBC3-SHA  DHE-RSA-AES128-SHA  DHE-DSS-AES128-SHA  AES128-SHA  IDEA-CBC-SHA  DHE-DSS-RC4-SHA  RC4-SHA  RC4-MD5  EXP1024-DHE-DSS-DES-CBC-SHA  EXP1024-DES-CBC-SHA  EDH-RSA-DES-CBC-SHA  EDH-DSS-DES-CBC-SHA  DES-CBC-SHA  EXP1024-DHE-DSS-RC4-SHA  EXP1024-RC4-SHA  EXP1024-RC4-MD5  EXP-EDH-RSA-DES-CBC-SHA  EXP-EDH-DSS-DES-CBC-SHA  EXP-DES
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Configuring Security Features negotiation with “Server Hello Done” message. Step3: The IP phone sends session key information (encrypted by server’s public key) in the “Client Key Exchange” message. Step4: Server sends “Change Cipher Spec” message to activate the negotiated options for all future messages it will send. IP phones can encrypt SIP with TLS, which is called SIPS.
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Administrator’s Guide for CP860 IP conference phones Procedure Configuration changes can be performed using the configuration files or locally. Configure TLS. .cfg Parameter: account.X.transport Configure the trusted certificates feature. Parameters: security.trust_certificates security.ca_cert security.cn_validation Configuration File Configure the server certificates feature. y000000000037.cfg Parameters: security.dev_cert Upload the trusted certificates. Parameter: trusted_certificates.
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Configuring Security Features et?p=server-cert&q=load Details of Configuration Parameters: Parameters account.X.transport (X = 1) Permitted Values Default Integer 0 0 or 1 1 Description: Configures the type of transport protocol. 0-UDP 1-TCP 2-TLS 3-DNS-NAPTR Web User Interface: Account->Register->Transport Phone User Interface: None security.trust_certificates Description: Enables or disables the IP phone to only trust the server certificates in the Trusted Certificates list.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default 0-Default certificates 1-Custom certificates 2-All certificates Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Security->Trusted Certificates->CA Certificates Phone User Interface: None security.
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Configuring Security Features Parameters Permitted Values trusted_certificates.url URL within 511 characters Default Blank Description: Configures the access URL of the custom trusted certificate used to authenticate the connecting server. Example: trusted_certificates.url = http://192.168.1.20/tc.crt Note: The certificate you want to upload must be in *.pem, *.crt, *.cer or *.der format.
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Administrator’s Guide for CP860 IP conference phones 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 phone. To configure TLS via web user interface: 312 1. Click on Account. 2. Select TLS from the pull-down list of the Transport.
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Configuring Security Features 3. Click Confirm to accept the change. To upload a trusted certificate via web user interface: 1. Click on Security->Trusted Certificates. 2. Click Browse to locate 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.
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Administrator’s Guide for CP860 IP conference phones 2. Click Browse to locate 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) encrypts RTP streams during VoIP phone calls to avoid interception and eavesdropping. The parties participating in the call must enable SRTP simultaneously.
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Configuring Security Features The callee receives the INVITE message with the RTP encryption algorithm, and then answers the call by responding with a 200 OK message which carries the negotiated RTP encryption algorithm.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default 1-Optional 2-Compulsory If it is set to 1 (Optional), the IP phone will negotiate with the other IP phone what type of encryption to utilize for the session. If it is set to 2 (Compulsory), the IP phone is forced to use SRTP during a call. Web User Interface: Account->Advanced->RTP Encryption (SRTP) Phone User Interface: None To configure SRTP via web user interface: 1. Click on Account-> Advanced. 2.
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Configuring Security Features same or different keys for configuration files) and generates encrypted configuration files with the same file name as before. This tool also encrypts the plaintext 16-character symmetric keys using a fixed key, which is the same as the one built in the IP phone, and generates new files named as .enc (xx indicates the name of the configuration file, for example, y000000000037_Security.enc for y000000000037.cfg file).
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Administrator’s Guide for CP860 IP conference phones automatically in the directory where the application tool is located. 2. Click Browse to locate configuration file(s) (e.g., y000000000037.cfg) from your local system in the Select File(s) field. To select multiple configuration files, you can select the first file and then press and hold the Ctrl key and select the next files. 3. (Optional.) Click Browse to locate the target directory from your local system in the Target Directory field.
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Configuring Security Features 6. Click OK. The target directory will be automatically opened. You can find the encrypted configuration file(s), encrypted key file(s) and an Aeskey.txt file storing plaintext AES key(s). Procedure Encryption method and AES keys can be configured using the configuration files or locally. Configure the decryption method and AES keys. Parameters： Configuration File y000000000037.cfg auto_provision.aes_key_in_file auto_provision.aes_key_16.com auto_provision.aes_key_16.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default If it is set to 1 (Enabled), the IP phone will download y000000000037_Security.enc and .enc files during auto provisioning, and then decrypts these files into the plaintext keys (e.g., key2, key3) respectively using the phone built-in key (e.g., key1). The IP phone then decrypts the encrypted configuration files using corresponding key (e.g., key2, key3).
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Configuring Security Features Parameters Permitted Values Default 0 or 1 0 Phone User Interface: None auto_provision.update_file_mode Description: Enables or disables the IP phone to update encrypted configuration settings only during auto provisioning. 0-Disabled 1-Enabled Web User Interface: None Phone User Interface: None To configure the AES keys via web user interface: 1. Click on Settings->Auto Provision. 2. Enter the values in the Common AES Key and MAC-Oriented AES Key fields. 3.
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Administrator’s Guide for CP860 IP conference phones 322
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Resource Files When configuring particular features, you may need to upload resource files (e.g., local contact directory, remote phone book) to the IP phone. The resources files can be local contact directory, remote phone book and so on. Ask Yealink field application engineer for resource file templates. If the resource file is to be used for all IP phones of the same model, the resource file access URL is best specified in the y000000000037.cfg file.
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Administrator’s Guide for CP860 IP conference phones The replace rule template helps with the creation of multiple replace rules. After setup, place the replace rule file to the provisioning server and specify the access URL of the file in the configuration files. When editing a replace rule template file, learn the following:  indicates the start of the template file and indicates the end of the template file.  Create replace rules between and .
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Resource Files The dial-now template helps with the creation of multiple dial-now rules. After setup, place the dial-now file to the provisioning server and specify the access URL of the file in the configuration files. When editing a dial-now template, learn the following:  indicates the start of a template and indicates the end of a template.  Create dial-now rules between and .  At most 100 dial-now rules can be added to the IP phone.
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Administrator’s Guide for CP860 IP conference phones The softkey layout template allows assigning different soft key layouts to different call states. The call states include CallFailed, CallIn, Connecting, Dialing, RingBack and Talking. After setup, place the softkey layout file to the provisioning server and specify the access URL of the file in the configuration files.
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Resource Files Directory provides easy access to frequently used lists. Users can access lists by pressing the Directory soft key when the IP phone is idle. The lists may contain Local Directory, History, Remote Phone Book and LDAP.
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Administrator’s Guide for CP860 IP conference phones Procedure Use the following procedures to customize a directory template. Customizing a directory template: 1. Open the template file using an ASCII editor. 2. For each directory list that you want to configure, edit the corresponding string in the file.
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Resource Files When editing a super search template, learn the following:  indicates the start of a template and indicates the end of a template.  The default display names of directory lists are Local Directory, History, Remote Phone Book and LDAP.  When specifying the priority of search results, the valid values are 1, 2, 3 and 4. 1 is the highest priority, 4 is the lowest.
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Administrator’s Guide for CP860 IP conference phones You can add contacts one by one on the IP phone directly. You can also add multiple contacts at a time and/or share contacts between IP phones using the local contact template file (Yealink-supplied template file is named as contact.xml). After setup, place the local contact file to the provisioning server, and specify the access URL of the file in the configuration files.
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Resource Files ring=”” specifies the ring tone for this contact. If it is left blank, the ring tone of the contact will be specified as Auto. group_id_name=”” specifies the existing group you want to add the contact to. 4. Specify the values within double quotes. 5. Save the change and place this file to the provisioning server. The following shows an example of a local contact file:
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Administrator’s Guide for CP860 IP conference phones Procedure Use the following procedures to customize an XML phone book. To customize a Menu.xml file: 1. Open the template file using an ASCII editor. 2. For each department that you want to add, add the following strings to the file. Each starts on a separate line: Department1 http://10.3.6.117:8080/Department1.xml Where: Specify the name of a department between and .
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Resource Files # http://10.3.6.117:8080/TextMenu When creating a Department.xml file, learn the following:  indicates the start of a department file and indicates the end of a department file.  Create contact lists for a department between and . To customize a Department.xml file: 1. Open the template file using an ASCII editor. 2.
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Administrator’s Guide for CP860 IP conference phones The following shows an example of a Department.xml file: Jack 1003 John 1004 Marry 1005 Note 334 Yealink supplies a phone book generation tool to quickly generate a remote XML phone book.
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Troubleshooting This chapter provides an administrator with general information for troubleshooting some common problems that he (or she) may encounter while using CP860 IP conference phones. IP phones can provide feedback in a variety of forms such as log files, packets, status indicators and so on, which can help an administrator more easily find the system problems and fix them. The following are helpful for better understanding and resolving the working status of the IP phone.
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Administrator’s Guide for CP860 IP conference phones 5: normal but significant condition 6: informational Procedure Log setting can be configured using the configuration files or locally. Configures the syslog mode. Parameters: syslog.mode Configures the IP address or domain name of the syslog server where to Configuration File y000000000037.cfg export the log files. Parameters: syslog.server Configures the severity level of the logs to be reported to a log file. Parameters: syslog.
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Troubleshooting Parameters Permitted Values Default Configures the IP phone to export log files to a syslog server or the local system. 0-Local 1-Server Note: If you change this parameter, the IP phone will reboot to make the change take effect. Web User Interface: Settings->Configuration->Export System Log Phone User Interface: None syslog.server IP address or domain name Blank Description: Configures the IP address or domain name of the syslog server when exporting log to the syslog server.
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Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Web User Interface: Settings->Configuration->System Log Level Phone User Interface: None To configure the system log level via web user interface: 1. Click on Settings->Configuration. 2. Select 6 from the pull-down list of System Log Level. 3. Click Confirm to accept the change. A dialog box pops up to prompt “Do you want to restart your machine?”. The configuration will take effect after a reboot. 4.
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Troubleshooting 3. Enter the IP address or domain name of the syslog server in the Server Name field. 4. Click Confirm to accept the change. A dialog box pops up to prompt “Do you want to restart your machine?”. The configuration will take effect after a reboot. 5. Click OK to reboot the phone. The system log will be exported successfully to the desired syslog server after a reboot. 6. Reproduce the issue. To export a log file to the local system via web user interface: 1.
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Administrator’s Guide for CP860 IP conference phones 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 purpose. To capture packets via web user interface: 340 1. Click on Settings->Configuration. 2. Click Start to start capturing signal traffic. 3. Reproduce the issue to get stack traces. 4. Click Stop to stop capturing.
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Troubleshooting 5. Click Export to open the file download window, and then save the file to your local system. To capture packets using the Ethernet software: Connect the Internet port of the IP phone and the PC to the same HUB, and then use Sniffer, Ethereal or Wireshark software to capture the signal traffic. The IP phone provides a troubleshooting feature called “Watch Dog”, which helps you monitor the IP phone status and provides the ability to get stack traces from the last time the IP phone failed.
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Administrator’s Guide for CP860 IP conference phones Details of the Configuration Parameter: Parameter Permitted Values Default 0 or 1 1 watch_dog.enable Description : Enables or disables Watch Dog feature. 0-Disabled 1-Enabled If it is set to 1 (Enabled), the IP phone will reboot automatically when the system is broken down. Web User Interface: Settings->Preference->Watch Dog Phone User Interface: None To configure watch dog via web user interface: 1. Click on Settings->Preference. 2.
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Troubleshooting Unavailable” and the icon  will appear on the LCD screen. If an active call on the IP phone is muted, LED indicators illuminate solid red. For more information on the icons, refer to Reading Icons on page 13. Wrong configurations may have an impact on your phone use. You can export configuration file to check the current configuration of the IP phone and troubleshoot if necessary. To export configuration file via web user interface: 1. Click on Settings->Configuration. 2.
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Administrator’s Guide for CP860 IP conference phones  If your phone is PoE powered, ensure that you are using a PoE-compliant switch or hub. ’ Do one of the following:  Ensure that the Ethernet cable is plugged into the Internet port on the IP phone and the Ethernet cable is not loose.  Ensure that the Ethernet cable is not damaged.  Ensure that the IP address and related network parameters are set correctly.  Ensure that your network switch or hub is operational.
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Troubleshooting If you have poor sound quality/acoustics like intermittent voice, low volume, echo or other noise, the possible reasons could be:  Users are seated too far out of recommended microphone range and sound faint, or are seated too close to sensitive microphones and cause echo.  Intermittent voice is mainly caused by packet loss, due to network congestion, and jitter, due to message recombination of transmission or receiving equipment (e.g.
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Administrator’s Guide for CP860 IP conference phones From: sip:sipsak@ CSeq: 10 NOTIFY Call-ID: 1234@ Event: check-sync;reboot=true The IP phone only uses logo file in DOB format, as the DOB format file has a high compression ratio (the size of the uncompressed file compared to that of the compressed file) and can be stored in smaller space. Tools for converting BMP format to DOB format are available.
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Troubleshooting ’ Do one of the following:  Ensure that the configuration is set correctly.  Reboot the phone. Some configurations require a reboot to take effect.  Ensure that the configuration is applicable to the IP phone model.  The configuration may depend on support from a server. “ ” “ ” They are codes that the IP phone sends to the server when a certain action takes place.
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Administrator’s Guide for CP860 IP conference phones The web user interface prompts the message “Do you want to reset to factory?”. 3. Click OK to confirm the resetting. The IP phone will be reset to factory sucessfully after startup. Note Reset of the phone may take a few minutes. Do not power off until the IP phone starts up successfully. Factory reset can restore the original password. All custom settings will be overwritten after reset.
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Appendix 802.1x — an IEEE Standard for port-based Network Access Control (PNAC). It is a part of the IEEE 802.1 group of networking protocols. It offers an authentication mechanism for devices to connect to a LAN or WLAN. ACS (Auto Configuration server) — responsible for auto-configuration of the Central Processing Element (CPE).
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Administrator’s Guide for CP860 IP conference phones 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. OID (Object Identifier) — assigned to an individual object within a MIB.
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Appendix Time Zone Time Zone Name −11:00 Samoa −10:00 United States-Hawaii-Aleutian −10:00 United States-Alaska-Aleutian −09:00 United States-Alaska Time −08:00 Canada(Vancouver, Whitehorse) −08:00 Mexico(Tijuana, Mexicali) −08:00 United States-Pacific Time −07:00 Canada(Edmonton, Calgary) −07:00 Mexico(Mazatlan, Chihuahua) −07:00 United States-Mountain Time −07:00 United States-MST no DST −06:00 Canada-Manitoba(Winnipeg) −06:00 Chile(Easter Islands) −06:00 Mexico(Mexico City,
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Administrator’s Guide for CP860 IP conference phones Time Zone 352 Time Zone Name 0 United Kingdom(London) 0 Morocco +01:00 Albania(Tirane) +01:00 Austria(Vienna) +01:00 Belgium(Brussels) +01:00 Caicos +01:00 Chad +01:00 Spain(Madrid) +01:00 Croatia(Zagreb) +01:00 Czech Republic(Prague) +01:00 Denmark(Kopenhagen) +01:00 France(Paris) +01:00 Germany(Berlin) +01:00 Hungary(Budapest) +01:00 Italy(Rome) +01:00 Luxembourg(Luxembourg) +01:00 Macedonia(Skopje) +01:00 Netherla
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Appendix Time Zone Time Zone Name +04:30 Afghanistan +05:00 Kazakhstan(Aqtobe) +05:00 Kyrgyzstan(Bishkek) +05:00 Pakistan(Islamabad) +05:00 Russia(Chelyabinsk) +05:30 India(Calcutta) +06:00 Kazakhstan(Astana, Almaty) +06:00 Russia(Novosibirsk, Omsk) +07:00 Russia(Krasnoyarsk) +07:00 Thailand(Bangkok) +08:00 China(Beijing) +08:00 Singapore(Singapore) +08:00 Australia(Perth) +09:00 Korea(Seoul) +09:00 Japan(Tokyo) +09:30 Australia(Adelaide) +09:30 Australia(Darwin) +10:00
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Administrator’s Guide for CP860 IP conference phones Valid types are: Format  N/A  Forward  DND  Call Return  Intercom  XML Group  Multicast Paging  History  Menu  Status  LDAP  Prefix  Local Directory  Local Group  XML Directory  Keypad Lock  Directory Integer when x=1, the default value is 28. when x=2, the default value is 61. when x=3, the default value is 5. Default Value when x=4, the default value is 30. when x=5, the default value is 28.
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Appendix 45-Local Group 47-XML Directory 50-Keypad Lock 61-Directory Example programablekey.1.type = 0 Parameter- Configuration File programablekey.X.value y000000000037.cfg (X=1-6, 9, 13) Description Configures the value for some key features. Format String Default Value Blank Range String within 99 characters When you assign the Prefix to the key, this Example parameter is used to add a specified prefix number before the dialed number. programablekey.1.
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Administrator’s Guide for CP860 IP conference phones This parameter is only applicable to Local Group/XML Group features. When the key feature is configured as Local Group, valid values are: 0-All contacts 1-First local group … 48-Forty-eighth local group When the key feature is configured as XML Group (remote phone book), valid values are: 0-First XML group 1-Second XML group … 4-Fifth XML group Format Integer Default Value 0 Range 0 to 48 Configures the second remote phone Example book.
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Appendix The following RFC’s and Internet drafts are supported:  RFC 1321—The MD5 Message-Digest Algorithm  RFC 1889—RTP Media control  RFC 2112—Multipart MIME  RFC 2246—The TLS Protocol Version 1.0  RFC 2327—SDP: Session Description Protocol  RFC 2543—SIP: Session Initiation Protocol  RFC 2616—Hypertext Transfer Protocol -- HTTP/1.
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Administrator’s Guide for CP860 IP conference phones  RFC 3428—Session Initiation Protocol (SIP) Extension for Instant Messaging  RFC 3455—Private Header (P-Header) Extensions to the SIP for the 3GPP  RFC 3486—Compressing the Session Initiation Protocol (SIP)  RFC 3489—STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)  RFC 3515—The Session Initiation Protocol (SIP) Refer Method  RFC 3550—RTP , RTCP, IETF RFC 3550  RFC 3556—Session Descrip
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Appendix  RFC 4317—Session Description Protocol (SDP) Offer/Answer Examples  RFC 4353—A Framework for Conferencing with the SIP  RFC 4475—Session Initiation Protocol (SIP) Torture  RFC 4485—Guidelines for Authors of Extensions to the SIP  RFC 4504—SIP Telephony Device Requirements and Configuration  RFC 4566—SDP: Session Description Protocol.
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Administrator’s Guide for CP860 IP conference phones Method Supported Notes INVITE that contains an existing Call-ID.
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Appendix Method Supported 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 Subscription-State Yes Supported Yes To Yes User-Agent Yes Via Yes Notes The following SIP responses are supported: 1xx Response—Informatio
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Administrator’s Guide for CP860 IP conference phones 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.
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Appendix 4xx Response Supported 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 Unavailable Yes 481 Call/Transaction Does Not Exist Notes Yes 482 Loop Detected Yes 483 Too Many Hops No 484 Address Incomplete Yes 485 Ambiguous No 486 Busy Here Yes 487 Request Terminated Yes 488 Not Accepta
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Administrator’s Guide for CP860 IP conference phones 6xx Response—Global Responses 6xx Response Supported 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 Yes Yes a—Media attribute Yes c—Connection information Yes m—Media name and transport address Yes s—Session name Yes t—Active time Yes SIP uses six request methods:  INVITE—Indicates a user is being in
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Appendix  SIP 2xx—Successful Responses  SIP 3xx—Redirection Responses  SIP 4xx—Client Failure Responses  SIP 5xx—Server Failure Responses  SIP 6xx—Global Failure Responses The following figure illustrates the scenario of a successful call. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones. The call flow scenario is as follows: 1. User A calls User B. 2. User B answers the call. 3. User B hangs up.
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Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends a SIP INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request:  The IP address of User B is inserted in the Request-URI field.  User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy  Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
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Appendix Step Action Description User B sends a SIP 200 OK response to F7 200 OK— User B to Proxy the proxy server. The 200 OK response Server notifies User A that the connection has been made. The proxy server forwards the 200 OK F8 200OK—Proxy Server to User 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. The ACK confirms that User A has received the 200 OK response.
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Administrator’s Guide for CP860 IP conference phones The call flow scenario is as follows: 1. User A calls User B. 2. User B is busy on the IP phone and unable or unwilling to take another call. The call cannot be set up successfully. User A Proxy Server User B F1. INVITE B F2. INVITE B F3. 100 Trying F4. 100 Trying F5. 486 Busy Here F6. 486 Busy Here F7. ACK F8. ACK Step Action Description User A sends the INVITE message to a proxy server.
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Appendix Step Action Description is specified. F2 INVITE—Proxy Server to User B The proxy server maps the SIP URI in the To field to User B. Proxy server forwards the INVITE message to User B. User B sends a SIP 100 Trying response F3 100 Trying—User B to Proxy to the proxy server. The 100 Trying Server response indicates that the INVITE request has been received by User B.
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Administrator’s Guide for CP860 IP conference phones The following figure illustrates the scenario of an unsuccessful call caused by the called user’s no answering. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones. The call flow scenario is as follows: 1. User A calls User B. 2. User B does not answer the call. 3. User A hangs up. The call cannot be set up successfully. User A Proxy Server User B F1. INVITE B F2. INVITE B F3.
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Appendix Step Action Description  The transaction number within a single call leg is identified in the CSeq field.  The media capability User A is ready to receive is specified.  The port on which User B is prepared to receive the RTP data is specified. F2 F3 INVITE—Proxy Server to User B The proxy server maps the SIP URI in the To field to User B. Proxy server forwards the INVITE message to User B.
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Administrator’s Guide for CP860 IP conference phones The following figure illustrates a successful call setup and call hold. In this scenario, the two end users are User A and User B. User A and User B are located at Yealink SIP IP phones. 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.
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Appendix Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request:  The IP address of User B is inserted 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.
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Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server. The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
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Appendix 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. 200 OK F316 200 OK F17. ACK F18.
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Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request:  The IP address of User B is inserted in the Request-URI field.  User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy  Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
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Appendix Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server, The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active. User C sends a SIP INVITE message to the proxy server.
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Administrator’s Guide for CP860 IP conference phones 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.
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Appendix The following figure illustrates a successful call between Yealink SIP IP phones in which two parties are in a call and then one of the parties transfers the call to a third party without consultation. This is called a blind transfer. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: 1. User A calls User B. 2. User B answers the call. 3.
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Administrator’s Guide for CP860 IP conference phones 4. User C answers the call. Call is established between User A and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6. 200 OK F7. ACK F8. ACK 2-way RTP channel established F9. REFER F10. 202 Accepted F11. REFER F12. 202 Accepted F17. BYE F18. BYE F19. 200 OK F20. 200 OK F21. INVITE C F22. INVITE C F23. 180 Ringing F24. 180 Ringing F25. 200 OK F26. 200 OK F27. ACK F28.
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Appendix Step Action Description User A sends an INVITE message to the proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request:  The IP address of User B is inserted 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.
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Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server, The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
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Appendix 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.
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Administrator’s Guide for CP860 IP conference phones 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.
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Appendix Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request:  The IP address of User B is inserted 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.
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Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends a SIP ACK to the proxy F7 ACK—User A to Proxy Server server, The ACK confirms that User A has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to F8 ACK—Proxy Server to User B User B. The ACK confirms that the proxy server has received the 200 OK response. The call session is now active.
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Appendix 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. User C sends a SIP 200 OK response to F19 200OK—User C to Proxy the proxy server.
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Administrator’s Guide for CP860 IP conference phones Step Action Description response indicates that User B accepts the transfer. User A terminates the call session by F27 BYE—User A to Proxy Server sending a SIP BYE request to the proxy server. The BYE request indicates that User A wants to release the call. F28 BYE—Proxy Server to User B The proxy server forwards the BYE request to User B. User B sends a SIP 200 OK response to F29 200OK—User B to Proxy the proxy server.
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Appendix 4. User C answers the call. Call is established between User A and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 302 Move Temporarily F4. ACK F5. 302 Move Temporarily F6. ACK F7. INVITE C F8. INVITE C F9. 180 Ringing F10. 180 Ringing F11. 200 OK F12. 200 OK F13. ACK F14.
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Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends an INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request:  The IP address of the User B is inserted in the Request-URI field.  User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy  Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
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Appendix 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. User C sends a SIP 180 Ringing F9 180 Ringing—User C to Proxy response to the proxy server.
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Administrator’s Guide for CP860 IP conference phones 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.
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Appendix Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request:  The IP address of User B is inserted 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.
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Administrator’s Guide for CP860 IP conference phones Step Action Description ACK message. F7 302 Move Temporarily—Proxy The proxy server forwards the 302 Server to User A Moved Temporarily message to User A. User A sends a SIP ACK to the proxy F8 ACK—User A to Proxy Server server. The ACK message notifies the proxy server that User A has received the ACK message. User A sends a SIP INVITE request to the F9 INVITE—User A to Proxy Server proxy server.
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Appendix 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.
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Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request:  The IP address of User B is inserted in the Request-URI field.  User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy  Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
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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.
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Administrator’s Guide for CP860 IP conference phones The following figure illustrates successful 3-way calling between Yealink CP860 IP conference 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: 398 1. User A calls User B. 2.
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Appendix 6. User A mixes the RTP channels and establishes a conference between User B and User C. User A User B Proxy Server F1. INVITE B F4. 180 Ringing F6. 200 OK F7. ACK User C F2. INVITE B F3. 180 Ringing F5. 200 OK F8. ACK Session1 established between User A and User B is active F9. INVITE(sendonly) Initiate three party conference F10. INVITE (sendonly) F11. 200 OK F12. 200 OK F13. ACK F14. ACK Session 1 established between User A and User B is hold F15. INVITE C F16. INVITE C F17.
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Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends the INVITE message to a proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request:  The IP address of User B is inserted in the Request-URI field.  User A is identified as the call session initiator in the From field. F1 INVITE—User A to Proxy  Server A unique numeric identifier is assigned to the call and is inserted in the Call-ID field.
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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.
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Administrator’s Guide for CP860 IP conference phones 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.
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Index Numeric Call Log 180 Ring Workaround 126 802.
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Administrator’s Guide for CP860 IP conference phones H P H.
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Index Transfer on Conference Hang Up Transfer via DTMF 172 Transport Layer Security (TLS) Troubleshooting 152 305 335 Troubleshooting Methods 335 Troubleshooting Solutions 343 TR-069 Device Management 278 U Upgrading Firmware 32 Use Outbound Proxy in Dialog User Agent Client (UAC) 2 User Agent Server (UAS) 3 User Password 128 46 V Verifying Startup Viewing Log Files VLAN 13 335 255 Voice Activity Detection VoIP Principle VPN 299 1 259 W Web Server Type Web User Interface 43 15