Yealink CP860 Administrator's Manual
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Summary of Contents for Yealink CP860

  • Page 2 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.
  • Page 3 Do not dispose of WEEE as unsorted municipal waste and have to collect such WEEE separately. We are striving to improve our documentation quality and we appreciate your feedback. Email your opinions and comments to DocsFeedback@yealink.com.
  • Page 4 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.
  • Page 5: About This Guide

    BroadSoft features on the BroadWorks web portal and IP phones. For support or service, please contact your Yealink reseller or go to Yealink Technical Support online: http://www.yealink.com/Support.aspx. The information detailed in this guide is applicable to the firmware version 72 or higher.
  • Page 6 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”...
  • Page 7: Table Of Contents

    VoIP Principle ............................ 1 SIP Components..........................2 Introduction of CP860 IP Conference Phones ................3 Physical Features of CP860 IP Conference Phones ............... 4 Key Features of CP860 IP Conference Phones ............... 4 Getting Started ................. 7 Connecting the IP Phones ....................... 7 Connecting the Network and Power ..................
  • Page 8 Administrator’s Guide for CP860 IP conference phones Configuring Basic Features ............ 39 Contrast ............................40 Backlight ............................42 Web Server Type..........................43 User Password ..........................46 Administrator Password ........................ 48 Phone Lock ............................. 50 Time and Date ..........................54 Language ............................66 Loading Language Packs ......................
  • Page 9: Table Of Contents

    Table of Contents Transfer on Conference Hang Up ....................152 Directed Call Pickup ........................153 Group Call Pickup ........................157 Call Return ............................ 160 Calling Line Identification Presentation ..................162 Connected Line Identification Presentation ................164 DTMF ............................. 165 Suppress DTMF Display ......................169 Transfer via DTMF ........................
  • Page 10 Administrator’s Guide for CP860 IP conference phones Voice Activity Detection ....................... 299 Comfort Noise Generation ....................301 Jitter Buffer ..........................302 Configuring Security Features ..........305 Transport Layer Security ......................305 Secure Real-Time Transport Protocol ..................314 Encrypting Configuration Files ....................316 Resource Files ................323...
  • Page 11 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 ................349 Appendix A: Glossary .........................
  • Page 13: Product Overview

    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.
  • Page 14 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 ...
  • Page 15 CP860 IP conference phones also support advanced functionalities, including LDAP , Sever Redundancy and Network Conference. CP860 IP conference phones comply with the SIP standard (RFC 3261), and they can only be used within a network that supports this type of phone.
  • Page 16: Physical Features Of Cp860 Ip Conference Phones

    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...
  • Page 17 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.726, iLBC VAD, CNG, AEC, PLC, AJB, AGC Full-duplex speakerphone with AEC Built in microphone arrray, 360 degree vocie pickup...
  • Page 18 Administrator’s Guide for CP860 IP conference phones...
  • Page 19: Getting Started

    Configuring Basic Network Parameters  Upgrading Firmware  This section introduces how to install CP860 IP conference phones with the components in packaging contents. Connecting the Network and Power Connecting the Optional Extension Microphones Kit Connecting the Optional USB Flash Drive...
  • Page 20 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. Power over Ethernet With the included or a regular Ethernet cable, the CP860 IP conference phone can be powered from a PoE-compliant switch or hub.
  • Page 21 Important! Do not unplug or remove power to the phone while it is updating firmware and configurations. You can connect optional extension microphones to enhance the room coverage of the conference phone. The Yealink-provided extension microphone kit contains two extension microphones.
  • Page 22 Administrator’s Guide for CP860 IP conference phones To connect the extension microphones: 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.
  • Page 23 Getting Started To connect a PC or mobile device: 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.
  • Page 24 24. Contacting the auto provisioning server CP860 IP conference phones support the FTP , TFTP , HTTP , and HTTPS protocols for auto provisioning and are configured by default to use TFTP protocol. If IP phones are configured to obtain configurations from the TFTP server, they will connect to the TFTP server and download the configuration file(s) during startup.
  • Page 25 When the IP phone has successfully passed through these steps, it starts up properly and is ready for use. Icons associated with different features may appear on the LCD screen. The following table provides a description for each icon on CP860 IP conference phone models. Icon Description...
  • Page 26: Phone User Interface

    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...
  • Page 27 IP phone. The MAC-Oriented CFG file is named after the MAC address of the IP phone. For example, if the MAC address of a CP860 IP conference phone is 001565113af8, the name of the MAC-Oriented CFG file must be 001565113af8.cfg.
  • Page 28 Administrator’s Guide for CP860 IP conference phones configuration files (y000000000037.cfg and <MAC>.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...
  • Page 29 MAC-oriented configuration file will override the same one in the common configuration file. Yealink supplies configuration files for each phone model, which is delivered with the IP phone firmware. The configuration files, supplied with each firmware release, must be used with that release.
  • Page 30 Administrator’s Guide for CP860 IP conference phones Create new common configuration files by performing the following steps: Create y000000000037.cfg files by using the Common CFG file from the distribution as templates. Edit the parameters in the file as desired. Copy configuration files to the home directory of the provisioning server.
  • Page 31 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.
  • Page 32 Administrator’s Guide for CP860 IP conference phones Parameter DHCP Option Description Vendor Class Identify the vendor type. Identifier Identify a TFTP server when the 'sname' field TFTP Server in the DHCP header has been used for DHCP Name options. Identify a bootfile when the 'file' field in the...
  • Page 33 Getting Started Details of Configuration Parameters: Parameters Permitted Values Default network.internet_port.type 0 or 2 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.
  • Page 34 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.
  • Page 35 Getting Started In the IPv4 Config block, mark the DHCP radio box. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after reboot. Click OK to reboot the phone. To configure static DNS address when DHCP is used via web user interface: Click on Network->Basic.
  • Page 36 Administrator’s Guide for CP860 IP conference phones Enter the desired values in the Primary DNS and Secondary DNS fields. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after a reboot.
  • Page 37 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 network.ip_address_mode Configuration File <MAC>.cfg 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. Web User Interface Navigate to: Local...
  • Page 38 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)
  • Page 39 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.gateway IPv4 Address Blank Description: Configures the IPv4 default gateway 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.
  • Page 40 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default network.secondary_dns 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.
  • Page 41 Getting Started Click OK to reboot the phone. To configure a static IPv4 address via web user interface: Click on Network->Basic. In the IPv4 Config block, mark the Static IP Address radio box. Enter the IP address, subnet mask, default gateway, primary DNS and secondary DNS in the corresponding fields.
  • Page 42 Wrong network settings may result in inaccessibility of your phone and may also have an impact on your network performance. For more information on these parameters, contact your network administrator. The CP860 IP conference phone has Internet port only. There are three optional methods of transmission configuration for Internet port: Auto-negotiation ...
  • Page 43: Web User Interface

    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.
  • Page 44: Provisioning Server

    Manually, from the local system  Automatically, from the provisioning server  The associated firmware name of the CP860 IP conference phone is 37.x.0.x.rom (x is replaced by the actual firmware version). Note You can download the latest firmware online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 45 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: Click on Settings->Upgrade. Click Browse. Locate the firmware from the local system.
  • Page 46 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 auto_provision.weekly.enable Configuration File y000000000037.cfg auto_provision.weekly.begin_time auto_provision.weekly.end_time...
  • Page 47 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.
  • Page 48 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).
  • Page 49 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.
  • Page 50 Administrator’s Guide for CP860 IP conference phones Make the desired change. 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...
  • Page 51: Configuring Basic Features

    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 ...
  • Page 52: Transfer On Conference Hang Up

    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 ...
  • Page 53 Configuring Basic Features Configure the contrast of the Phone User Interface LCD screen. Details of Configuration Parameters: Parameters Permitted Values Default phone_setting.contrast Integer from 1 to 10 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.
  • Page 54: Backlight

    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.
  • Page 55: Web Server Type

    Configuring Basic Features To configure the backlight via web user interface: Click on Settings->Preference. Select the desired value from the pull-down list of Backlight Time (seconds). Click Confirm to accept the change. To configure the backlight via phone user interface: Press Menu->Settings->Basic Settings->Display->Backlight Settings.
  • Page 56 Administrator’s Guide for CP860 IP conference phones Specify the web access type, HTTP port and HTTPS port. Web User Interface Navigate to: Local http://<phoneIPAddress>/servl et?p=network-adv&q=load Phone User Interface Specify the web access type. Details of Configuration Parameters: Parameters Permitted Values Default wui.http_enable...
  • Page 57: Configure Via Web User Interface

    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.
  • Page 58: User Password

    Administrator’s Guide for CP860 IP conference phones The default HTTPS port is 443. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the phone.
  • Page 59 Configuring Basic Features Procedure User password can be changed using the configuration files or locally. Change the user password of the IP phone. Configuration File y000000000037.cfg Parameter: security.user_password Change the user password of the IP phone. Local Web User Interface Navigate to: http://<phoneIPAddress>/servlet ?p=security&q=load...
  • Page 60: Administrator Password

    Administrator’s Guide for CP860 IP conference phones Valid characters are ASCII characters 32-126(0x20-0x7E) except 58(3A). Click Confirm to accept the change. Note 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.
  • Page 61 Configuring Basic Features Details of the Configuration Parameter: Parameter Permitted Values Default security.user_password String within 32 characters 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.
  • Page 62: Phone Lock

    Administrator’s Guide for CP860 IP conference phones Enter a new administrator password in the New PWD field and Confirm PWD field. Valid characters are ASCII characters 32-126(0x20-0x7E). 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.
  • Page 63 Configuring Basic Features ures-phonelock&q=load Assign a keypad lock key. Navigate to: http://<phoneIPAddress>/servlet?p=dssk ey&model=2&q=load Configure the phone lock type. Phone User Interface Configure the unlock PIN. Details of Configuration Parameters: Parameters Permitted Values Default phone_setting.phone_lock.enable 0 or 1 Description: Enables or disables phone lock feature. 0-Disabled 1-Enabled Web User Interface:...
  • Page 64 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Lock->Keypad Lock Type phone_setting.phone_lock.unlock_pin characters within 15 digits 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.lock_time_out...
  • Page 65 Configuring Basic Features To configure phone lock via web user interface: Click on Features->Phone Lock. Select the desired type from the pull-down list of Keypad Lock Enable. Select the desired type from the pull-down list of Keypad Lock Type. Enter unlock PIN (numeric characters) in the phone Unlock PIN (0~15 Digit) field. Enter the desired time in the phone Lock Time Out (0~3600s) field.
  • Page 66: Time And Date

    Administrator’s Guide for CP860 IP conference phones Press the soft key to select the desired type from the Lock type field. Press the Save soft key to accept the change. To configure the unlock PIN via phone user interface: Press Menu->Settings->Basic Settings->Phone Unlock PIN.
  • Page 67 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.
  • Page 68 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.
  • Page 69 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 &...
  • Page 70 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...
  • Page 71 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 Description: Configures the DST time type. 0-By Date 1-By Week Note: It works only if the parameter “local_time.summer_time”...
  • Page 72 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->...
  • Page 73 Configuring Basic Features Parameters Permitted Values Default local_time.offset_time 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...
  • Page 74 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: Click on Settings->Time &...
  • Page 75 Configuring Basic Features Enter the domain names or IP addresses in the Primary Server and Secondary Server fields respectively. Enter the desired time interval in the Synchronism (15~86400s) field. 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.
  • Page 76 Administrator’s Guide for CP860 IP conference phones Enter the desired time in the End Hour of Day field. Enter the desired offset time in the Offset (minutes) field. Click Confirm to accept the change. To configure the time and date manually via web user interface: Click on Settings->Time &...
  • Page 77 Configuring Basic Features Select the desired value from the pull-down list of Time Format. Select the desired value from the pull-down list of Date Format. Click Confirm to accept the change. To configure the NTP server and time zone via phone user interface: Press Menu->Settings->Basic Settings->Time &...
  • Page 78: Language

    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.
  • Page 79 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. Specify the access URL of the language pack.
  • Page 80 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.
  • Page 81 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.
  • Page 82: Logo Customization

    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.
  • Page 83 Configuring Basic Features Parameters Permitted Values Default 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 URL within 511 characters Blank Description: Configures the access URL of the custom logo file. Example:...
  • Page 84: Softkey Layout

    Administrator’s Guide for CP860 IP conference phones Click Browse to select the logo file from your local system. Click Upload to upload the file. Click Confirm to accept the change. The custom logo screen and the idle screen are displayed alternately.
  • Page 85 Configuring Basic Features Call State Default Soft Key Optional Soft Key DPickup Empty Empty Empty Switch RingBack Empty Cancel RingBack Transfer Empty Empty Switch SemiAttendTransBack Empty Cancel Transfer Empty Hold Mute Conference SWAP Cancel NewCall Switch Talk Answer Reject Start Record Pause Record Resume Record Stop Record...
  • Page 86 Administrator’s Guide for CP860 IP conference phones Call State Default Soft Key Optional Soft Key Resume Record Stop Record Transfer Empty Directory PreTrans Delete Switch Cancel Send Empty Empty Hold Switch Split Answer Cancel Reject Mute Conferenced Manager Start Record...
  • Page 87 Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default custom_softkey_call_failed.url 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”...
  • Page 88 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Web User Interface: None Phone User Interface: None custom_softkey_dialing.url 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 Dialing state.
  • Page 89 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: Click on Settings->Softkey Layout.
  • Page 90 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 Configure a key tone and send Configuration File y000000000037.cfg...
  • Page 91 Configuring Basic Features Parameters Permitted Values Default Phone User Interface: Menu->Features->Key as Send features.key_tone 0 or 1 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.
  • Page 92 Administrator’s Guide for CP860 IP conference phones Select the desired value from the pull-down list of Key As Send. Click Confirm to accept the change. To configure a key tone and send tone via web user interface: Click on Features->Audio.
  • Page 93: Dial Plan

    Configuring Basic Features Press the Save soft key to accept the change. Note 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”...
  • Page 94: Replace Rule

    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:...
  • Page 95 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 String within 32 characters Blank (X ranges from 1 to 100) Description: Configures the alternate number to replace the entered number.
  • Page 96: Dial-Now

    Administrator’s Guide for CP860 IP conference phones Enter the string in the Replace field. 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.
  • Page 97 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://<phoneIPAddress>/servlet ?p=settings-dialnow&q=load Local Web User Interface Configure the delay time for the dial-now rule. Navigate to: http://<phoneIPAddress>/servlet ?p=features-general&q=load Details of Configuration Parameters: Parameters...
  • Page 98 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 Blank Description: Configures the access URL of the dial-now rule template file. Example: dialplan_dialnow.url = http://192.168.10.25/dialnow.xml...
  • Page 99: Area Code

    Configuring Basic Features Enter the desired time within 1-14 (in seconds) in the Time-Out for Dial-Now Rule field. Click Confirm to accept the change. Area codes are also known as Numbering Plan Areas (NPAs). They usually indicate geographical areas in one country. When 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.
  • Page 100 Administrator’s Guide for CP860 IP conference phones Navigate to: http://<phoneIPAddress>/servlet ?p=settings-areacode&q=load Details of Configuration Parameters: Parameters Permitted Values Default dialplan.area_code.code String within 16 characters Blank Description: Configures the area code to be added before the entered numbers when dialing out.
  • Page 101: Block Out

    Configuring Basic Features Enter desired values in the Code, Min Length (1-15) and Max Length (1-15) fields. 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”.
  • Page 102: Hotline

    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: Click on Settings->Dial Plan->Block Out.
  • Page 103 Configuring Basic Features Procedure Hotline can be configured using the configuration files or locally. Configure the hotline number. Parameter: features.hotline_number Specify the time (in seconds) the Configuration File y000000000037.cfg IP phone waits to automatically dial out the hotline number. Parameter: features.hotline_delay Configure the hotline number.
  • Page 104 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.
  • Page 105: Directory

    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.
  • Page 106: Search Source List In Dialing

    Administrator’s Guide for CP860 IP conference phones click To adjust the display order of enabled lists, select the desired list and then click Click Confirm to accept the change. The IP phone LCD screen will display the enabled list(s) in the adjusted order.
  • Page 107 Configuring Basic Features ?p=contacts-favorite&q=load Details of the Configuration Parameter: Parameter Permitted Values Default super_search.url 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: Click on Directory->Setting.
  • Page 108: Call Log

    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.
  • Page 109: Missed Call Log

    Configuring Basic Features Select the desired value from the pull-down list of Save Call Log. Click Confirm to accept the change. To configure the call log via phone user interface: Press Menu-> Features-> History Setting. Press the soft key to select the desired value from the History Record field. Press the Save soft key to accept the change.
  • Page 110 Administrator’s Guide for CP860 IP conference phones http://<phoneIPAddress>/servlet ?p=account-basic&q=load&acc Details of the Configuration Parameter: Parameter Permitted Values Default account.X.missed_calllog 0 or 1 (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.
  • Page 111: Local Directory

    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.
  • Page 112 Administrator’s Guide for CP860 IP conference phones In the Group Setting block, enter the new group name in the Group field. Select the desired group ring tone from the pull-down list of Ring. Click Add to add the group. To add a contact to the local directory via web user interface: Click on Directory->Local Directory.
  • Page 113: Live Dialpad

    Configuring Basic Features Click Add to add the contact. To add a group to the local directory via phone user interface: Press Menu->Directory->Local Directory. Press the AddGrp soft key. Enter the desired group name in the Name field. Press the soft key to select the desired ring tone from the Ring Tones field.
  • Page 114 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.
  • Page 115: Call Waiting

    Configuring Basic Features Enter the desired delay time in the Inter Digit Time (1~14s) field. 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.
  • Page 116 Administrator’s Guide for CP860 IP conference phones Details of Configuration Parameters: Parameters Permitted Values Default call_waiting.enable 0 or 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.
  • Page 117 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:...
  • Page 118 Administrator’s Guide for CP860 IP conference phones (Optional.) Enter the call waiting off code in the Call Waiting Off Code field. Click Confirm to accept the change. To configure the call waiting tone via web user interface: Click on Features->Audio.
  • Page 119: Auto Redial

    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.
  • Page 120 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.
  • Page 121: Auto Answer

    Configuring Basic Features Enter the desired times in the Auto Redial Times (1~300) field. The default value is 10. Click Confirm to accept the change. To configure auto redial via phone user interface: Press Menu->Features->Auto Redial. Press the soft key to select Enable from the Auto Redial field. Enter the desired time in the Redial Interval field.
  • Page 122 Administrator’s Guide for CP860 IP conference phones account.X.auto_answer_mute_enable Specify a period of delay time for auto answer. y000000000037.cfg Parameter: features.auto_answer_delay Configure auto answer. Navigate to: http://<phoneIPAddress>/servlet?p=a ccount-basic&q=load&acc=0 Web User Interface Specify a period of delay time for auto Local answer.
  • Page 123 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 Integer from 1 to 4 (X = 1)
  • Page 124: Anonymous Call

    Administrator’s Guide for CP860 IP conference phones Enter the desired time (in seconds) in the Auto-Answer Delay (1~4s) field. Click Confirm to accept the change. To configure auto answer and auto answer mute via phone user interface: Press Menu->Features->Auto Answer.
  • Page 125 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: <sip:1012@10.2.1.199> 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.
  • Page 126 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) Description: Configures the IP phone to send anonymous on/off code to activate/deactivate the server-side anonymous call feature.
  • Page 127 Configuring Basic Features Parameters Permitted Values Default Menu->Features->Anonymous Call->On Code account.X.anonymous_call_offcode String within 32 characters Blank (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).
  • Page 128: Anonymous Call Rejection

    Administrator’s Guide for CP860 IP conference phones Press Menu->Features->Anonymous Call. Press the soft key to select Enable from the Local Anonymous field. (Optional.) Press the soft key to select the desired value from the Anonymous Code field. (Optional.) Enter the anonymous call on code in the On Code field.
  • Page 129 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”.
  • Page 130: Do Not Disturb

    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: Click on Account->Basic. Select the desired value from the pull-down list of Anonymous Call Rejection.
  • Page 131 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.
  • Page 132 Administrator’s Guide for CP860 IP conference phones Configure DND. Details of Configuration Parameters: Parameters Permitted Values Default features.dnd.enable 0 or 1 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.
  • Page 133 Configuring Basic Features Parameters Permitted Values Default Features->Forward& DND->DND->DND Off Code Phone User Interface: Menu->Features->DND->Off Code features.dnd_refuse_code 404, 480 or 486 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.
  • Page 134 Administrator’s Guide for CP860 IP conference phones In the desired programable key field, select DND from the pull-down list of Type. Click Confirm to accept the change. To configure the DND feature via web user interface: Click on Features->Forward & DND.
  • Page 135: Busy Tone Delay

    Configuring Basic Features Select the desired type from the pull-down list of Return Code When DND. 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.
  • Page 136: Return Code When Refuse

    Administrator’s Guide for CP860 IP conference phones Details of the Configuration Parameter: Parameter Permitted Values Default features.busy_tone_delay 0, 3 or 5 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.
  • Page 137 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 refusing a call.
  • Page 138: Early Media

    Administrator’s Guide for CP860 IP conference phones Select the desired value from the pull-down list of Return Code When Refuse. 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.
  • Page 139 Configuring Basic Features Details of the Configuration Parameter: Parameter Permitted Values Default phone_setting.is_deal180 0 or 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.
  • Page 140: Use Outbound Proxy In Dialog

    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.
  • Page 141: Sip Session Timer

    Configuring Basic Features Parameter Permitted Values Default None To specify whether to use outbound proxy server in a dialog via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Use Outbound Proxy in Dialog. Click Confirm to accept the change.
  • Page 142 Administrator’s Guide for CP860 IP conference phones account.X.advanced.timer_t4 Configure SIP session timer. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=account-adv&q=load&acc= Details of Configuration Parameters: Parameters Permitted Values Default account.X.advanced.timer_t1 Float from 0.5 to10 (X = 1) Description: Configures the SIP session timer T1 (in seconds).
  • Page 143: Call Hold

    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: Click on Account->Advanced.
  • Page 144 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.
  • Page 145 Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default features.play_hold_tone.enable 0 or 1 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...
  • Page 146 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Features->General Information->RFC 2543 Hold Phone User Interface: None account.X.music_server_uri SIP URI within 256 characters Blank (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, <sip:moh@sip.com>, <yealink.com>...
  • Page 147 Configuring Basic Features To configure call hold tone and call hold tone delay via web user interface: Click on Features->General Information. Select the desired value from the pull-down list of Play Hold Tone. Enter the desired time in the Play Hold Tone Delay field. Click Confirm to accept the change.
  • Page 148: Session Timer

    Administrator’s Guide for CP860 IP conference phones Enter the SIP URI (e.g., sip:moh@sip.com) in the Music Server URI field. 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.
  • Page 149 Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default account.X.session_timer.enable 0 or 1 (X = 1) 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.
  • Page 150 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default 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.
  • Page 151: Call Forward

    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.
  • Page 152 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://<phoneIPAddress>/servlet ?p=features-forward&q=load Web User Interface Configure forward international. Local Navigate to: http://<phoneIPAddress>/ servlet?p=features-general&q=l Configure call forward.
  • Page 153 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.
  • Page 154 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...
  • Page 155 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:...
  • Page 156 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...
  • Page 157 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 Description:...
  • Page 158 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). Click Confirm to accept the change. To configure the forward international feature via web user interface: Click on Features->General Information.
  • Page 159: Call Transfer

    Configuring Basic Features To enable call forward via phone user interface: Press Menu->Features->Call Forward. Press to select the desired forwarding type, and then press the Enter soft key. Depending on your selection: a.) If you select Always Forward: 1) Press the soft key to select Enable from the Always Forward field.
  • Page 160 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.
  • Page 161 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.on_hook_trans_enable...
  • Page 162: Network Conference

    Administrator’s Guide for CP860 IP conference phones Select the desired values from the pull-down lists of Semi-Attended Transfer, Blind Transfer On Hook and Semi-Attend Transfer On Hook. 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).
  • Page 163 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:...
  • Page 164 Administrator’s Guide for CP860 IP conference phones Enter the conference URI in the Conference URI field. 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.
  • Page 165 Configuring Basic Features Details of the Configuration Parameter: Parameter & Description Permitted Values Default transfer.tran_others_after_conf_enable 0 or 1 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...
  • Page 166 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.
  • Page 167 Configuring Basic Features Details of Configuration Parameters: Parameters Permitted Values Default 0 or 1 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...
  • Page 168 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: Click on Features->Call Pickup. Select the desired value from the pull-down list of Directed Call Pickup.
  • Page 169 Configuring Basic Features Enter the directed call pickup code in the Directed Call Pickup Code field. 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.
  • Page 170 Administrator’s Guide for CP860 IP conference phones code Configure the group call pickup feature on a phone basis. Navigate to: http://<phoneIPAddress>/servl et?p=features-callpickup&q=lo Configure the group call pickup code on a phone basis. Navigate to: Local Web User Interface http://<phoneIPAddress>/servl et?p=features-callpickup&q=lo Configure the group call pickup code on a per-line basis.
  • Page 171 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:...
  • Page 172 Administrator’s Guide for CP860 IP conference phones Enter the group call pickup code in the Group Call Pickup Code field. Click Confirm to accept the change. To configure the group call pickup code on a per-line basis via web user interface: Click on Account->Advanced.
  • Page 173 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. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet ?p=dsskey&model=2&q=load Details of Configuration Parameters: Parameter Permitted Values Default...
  • Page 174: Calling Line Identification Presentation

    Administrator’s Guide for CP860 IP conference phones In the desired programable key field, select Call Return from the pull-down list of Type. 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.
  • Page 175 Configuring Basic Features Details of the Configuration Parameter: Parameter Permitted Values Default account.X.cid_source 0, 1, 2, 3, 4 or 5 (X = 1) 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”...
  • Page 176: Connected Line Identification Presentation

    Administrator’s Guide for CP860 IP conference phones 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.
  • Page 177: Dtmf

    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.
  • Page 178 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.
  • Page 179 Configuring Basic Features RTP Event packet. Navigate to: http://<phoneIPAddress>/servl et?p=features-general&q=loa Details of Configuration Parameters: Parameters Permitted Values Default account.X.dtmf.type 0, 1, 2 or 3 (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.
  • Page 180 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...
  • Page 181: Suppress Dtmf Display

    Configuring Basic Features Enter the desired value in the DTMF Payload Type (96~127) field. Click Confirm to accept the change. To configure the number of times to send the end RTP Event packet via web user interface: Click on Features->General Information. Select the desired value (1-3) from the pull-down list of DTMF Repetition.
  • Page 182 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.
  • Page 183 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:...
  • Page 184: Transfer Via Dtmf

    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.
  • Page 185 Configuring Basic Features Parameters Permitted Values Default Description: the DTMF digits to be transmitted to perform call transfer. Valid values Configures 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).
  • Page 186 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.
  • Page 187: Intercom

    Configuring Basic Features Parameters Permitted Values Default programablekey.X.value String within 99 characters blank (X=1-6, 9, 13) Description: Configures the intercom number. Example: programablekey.2.value = 1008 Web User Interface: Programable ->Value DSSKey-> Phone User Interface: None To configure an intercom key via web user interface: Click on DSSKey->Programable Key.
  • Page 188: Incoming Intercom Calls

    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.
  • Page 189 Configuring Basic Features Parameters Permitted Values Default Web User Interface: Features->Intercom->Accept Intercom Phone User Interface: Menu->Features->Intercom->Accept Intercom features.intercom.mute 0 or 1 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.
  • Page 190 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.
  • Page 191: Configuring Advanced Features

    Configuring Advanced Features This chapter provides information for making configuration changes for the following advanced features: Distinctive Ring Tones  Tones  Remote Phone Book  LDAP  Message Waiting Indicator  Multicast Paging  Action URL  Action URI ...
  • Page 192 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.
  • Page 193 Configuring Advanced Features Minimum Nominal Maximum Bellcore Pattern Pattern Cadence Duration Duration Duration Tone (ms) (ms) (ms) Silent Ringing Long 1025 Silent 2975 4000 4400 Ringing Short Silent Ringing Long 1000 1100 Bellcore-dr4 Silent Ringing Short Silent 2975 4000 4400 Bellcore-dr5 Ringing Note...
  • Page 194: Auto Answer

    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.
  • Page 195 Configuring Advanced Features http://<phoneIPAddress>/servlet?p=accou nt-adv&q=load&acc=0 Configure the internal ringer text and internal ringer file. Navigate to: http://<phoneIPAddress>/servlet?p=setting s-ring&q=load Details of Configuration Parameters: Parameters Permitted Values Default account.X.alert_info_url_enable 0 or 1 (X = 1) Description: Enables or disables the IP phone to download the ring tone from the URL contained in the Alert-Info header.
  • Page 196 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...
  • Page 197 Configuring Advanced Features Select the desired value from the pull-down list of Distinctive Ring Tones. Click Confirm to accept the change. To configure the internal ringer text and internal ringer file via web user interface: Click on Settings->Ring. Enter the keywords in the Internal Ringer Text fields.
  • Page 198 Administrator’s Guide for CP860 IP conference phones Select the desired ring tones for each text from the pull-down lists of Internal Ringer File. 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.
  • Page 199 Configuring Advanced Features Great Britain  Greece  Hungary  Lithuania  India  Italy  Japan  Mexico  New Zealand  Netherlands  Norway  Portugal  Spain  Switzerland  Sweden  Russia  United States  Chile ...
  • Page 200 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 voice.tone.busy Configuration File y000000000037.cfg 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.
  • Page 201 Configuring Advanced Features Parameters Permitted Values Default Phone User Interface: None voice.tone.dial String Blank 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.
  • Page 202 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.
  • Page 203 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:...
  • Page 204 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.
  • Page 205 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.
  • Page 206 Administrator’s Guide for CP860 IP conference phones Details of Configuration Parameters: Parameters Permitted Values Default remote_phonebook.data.X.url 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...
  • Page 207 Configuring Advanced Features Parameters Permitted Values Default features.remote_phonebook.flash_time Integer from 120 to 2592000 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.
  • Page 208 Administrator’s Guide for CP860 IP conference phones Enter the desired time in the Search Flash Time (Seconds) field. 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.
  • Page 209 Mobile or cellular phone number ipPhone IPphoneNumber Home phone number LDAP Phonebook on Yealink IP Phones For more information on LDAP , refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Procedure LDAP can be configured using the configuration files or locally.
  • Page 210 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://<phoneIPAddress>/servl et?p=contacts-LDAP&q=load Local Web User Interface Assign an LDAP key. Navigate to: http://<phoneIPAddress>/servl et?p=dsskey&model=2&q=loa Details of Configuration Parameters: Parameters...
  • Page 211 Configuring Advanced Features Parameters Permitted Values Default Web User Interface: Directory->LDAP->LDAP Name Filter Phone User Interface: None ldap.number_filter String within 99 characters Blank 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.
  • Page 212 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Web User Interface: Directory->LDAP->Port Phone User Interface: None ldap.base String within 99 characters Blank 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.
  • Page 213 Configuring Advanced Features Parameters Permitted Values Default ldap.password =secret Web User Interface: Directory->LDAP->Password Phone User Interface: None ldap.max_hits Integer from 1 to 32000 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.
  • Page 214 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.
  • Page 215 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 Description: Enables or disables the IP phone to sort the search results in alphabetical order or numerical order.
  • Page 216 Administrator’s Guide for CP860 IP conference phones To configure LDAP via web user interface: Click on Directory->LDAP. Select Enabled from the pull-down list of Enable LDAP. Enter the values in the corresponding fields. Select the desired values from the corresponding pull-down lists.
  • Page 217 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.
  • Page 218 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 Integer from 0 to 84600 3600 (X = 1) Description: Configures MWI subscribe expiry time (in seconds).
  • Page 219 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: Click on Account->Advanced.
  • Page 220 Administrator’s Guide for CP860 IP conference phones Enter the desired voice number in the Voice Mail field. 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.
  • Page 221 Configuring Advanced Features Parameters: programablekey.X.type programablekey.X.value. Assign a multicast paging key. Navigate to: http://<phoneIPAddress>/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://<phoneIPAddress>/servlet ?p=features-general&q=load Details of the Configuration Parameter: Parameters Permitted Values Default...
  • Page 222 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Web User Interface: Programable Programable DSSKey-> Key-> KeyX->Type Phone User Interface: None programablekey.X.value String within 99 blank (X=1-6, 9, 13) characters Description: Configures the multicast IP address and port number.
  • Page 223 Configuring Advanced Features Select the desired codec from the pull-down list of Multicast Codec. 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.
  • Page 224 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 multicast.listen_address.X.ip_address Configuration y000000000037.cfg File Configure the Paging Barge and Paging Priority Active features. Parameters: multicast.receive_priority.enable...
  • Page 225 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 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.
  • Page 226 Administrator’s Guide for CP860 IP conference phones Enter the listening multicast address and port number in the Listening Address field. 1 is the highest priority and 10 is the lowest priority. Enter the label in the Label field. The label will appear on the LCD screen when receiving the RTP multicast.
  • Page 227 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).
  • Page 228 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.
  • Page 229 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 call. $remote The SIP URI of the caller when the IP phone receives an incoming call.
  • Page 230 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.transfer_finished action_url.transfer_failed Configure the action URL. Navigate to: Local Web User Interface http://<phoneIPAddress>/servlet?p=fea tures-actionurl&q=load Details of Configuration Parameters:...
  • Page 231 Configuring Advanced Features Parameters Permitted Values Default $active_url  $active_user  $active_host  $local  $remote  $display_local  $display_remote  $call_id  Example: action_url. setup_completed = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Setup Completed action_url.registered Blank URL within 511 characters Description: Configures the action URL the IP phone sends after an account is registered.
  • Page 232 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=$ip Web User Interface: Features->Action URL->Register Failed Phone User Interface: None action_url.off_hook URL within 511 characters...
  • Page 233 Configuring Advanced Features Parameters Permitted Values Default action_url.incoming_call = http://192.168.0.20/help.xml?IP=$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.xml?IP=$ip Web User Interface: Features->Action URL->Outgoing Call...
  • Page 234 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Phone User Interface: None action_url.dnd_off URL within 511 characters Blank Description: Configures the action URL the IP phone sends when DND feature is disabled. Example: action_url.dnd_off = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Close DND...
  • Page 235 Configuring Advanced Features Parameters Permitted Values Default action_url.busy_fwd_on URL within 511 characters 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=$ip Web User Interface: Features->Action URL->Open Busy Forward Phone User Interface: None action_url.busy_fwd_off URL within 511 characters...
  • Page 236 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Configures the action URL the IP phone sends when no answer forward feature is disabled. Example: action_url.no_answer_fwd_off = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Close No Answer Forward Phone User Interface: None action_url.transfer_call...
  • Page 237 Configuring Advanced Features Parameters Permitted Values Default Example: action_url.attended_transfer_call = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Attended Transfer Phone User Interface: None action_url.hold URL within 511 characters Blank Description: Configures the action URL the IP phone sends when placing a call on hold. Example: action_url.hold = http://192.168.0.20/help.xml?IP=$ip Web User Interface:...
  • Page 238 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Features->Action URL->Mute Phone User Interface: None action_url.unmute Blank URL within 511 characters 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=$ip Web User Interface: Features->Action URL->UnMute...
  • Page 239 Configuring Advanced Features Parameters Permitted Values Default action_url.busy_to_idle URL within 511 characters Blank Description: Configures the action URL the IP phone sends when changing the state of the IP phone from busy to idle. Example: action_url.busy_to_idle = http://192.168.0.20/help.xml?IP=$ip Web User Interface: Features->Action URL->Busy To Idle Phone User Interface: None...
  • Page 240 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=$ip Web User Interface: Features->Action URL->Forward Incoming Call Phone User Interface: None action_url.reject_incoming_call...
  • Page 241 Configuring Advanced Features Parameters Permitted Values Default action_url.transfer_finished = http://192.168.0.20/help.xml?IP=$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.xml?IP=$ip Web User Interface:...
  • Page 242 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.
  • Page 243 Configuring Advanced Features Variable Value Phone Action BTrans=xxx Perform a blind transfer to xxx. CALLEND End a call. Note 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&”...
  • Page 244 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.
  • Page 245 Working Server: Server 1 is configured with the domain name of the working server. For example, yealink.pbx.com. DNS mechanism is used such that the working server is resolved to multiple physical SIP servers for failover purpose. The working server is deployed in redundant pairs, designated as primary and secondary servers.
  • Page 246: Phone Registration

    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).
  • Page 247 Blank characters (X = 1, Y ranges from 1 to 2) 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.port...
  • Page 248 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default account.X.sip_server.Y.retry_counts Integer from 0 to 20 (X = 1, Y ranges from 1 to 2) 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.
  • Page 249 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.
  • Page 250 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: Click on Account->Register. Configure registration parameters of the account in the corresponding fields.
  • Page 251 DNS-NAPTR, NAPTR and SRV queries will be tried before falling to A query. If no port is found through the DNS query, 5060 will be used. The following details the procedures of DNS query for the IP phone to resolve the domain name (e.g., yealink.pbx.com) of working server into the IP address, port and transport protocol.
  • Page 252 SRV query next. TCP will be used, targeted to a host determined by an SRV query of “_sip._tcp.yealink.pbx.com”. If the flag of the NAPTR record returned is empty, the IP phone will perform the NAPTR query again according to the previous NAPTR query result.
  • Page 253 The two records also contain a port “5060”, the IP phone uses this port. If the Target is not a numeric IP address, the IP phone performs the A query. So in this case, the IP phone uses “server1.yealink.pbx.com” and “server2.yealink.pbx.com" for the A query.
  • Page 254 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.
  • Page 255 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:...
  • Page 256 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.priority account.X.dns_cache_srv.Y.target Configuration File <MAC>.cfg account.X.dns_cache_srv.Y.weight account.X.dns_cache_srv.Y.ttl account.X.dns_cache_a.Y.name account.X.dns_cache_a.Y.ip...
  • Page 257 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 Blank (X= 1, Y ranges from 1 to 12) 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).
  • Page 258 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:...
  • Page 259 Domain name Blank (X= 1, Y ranges from 1 to 12) Description: Configures the domain name in SRV record Y. Example: account.1.dns_cache_srv.1.name = _sip._tcp.yealink.pbx.com Web User Interface: None Phone User Interface: None account.X.dns_cache_srv.Y.port Integer from 0 to 65535 (X= 1, Y ranges from 1 to 12) Description: Configures the port to be used in SRV record Y.
  • Page 260 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Phone User Interface: None account.X.dns_cache_srv.Y.priority Integer from 0 to 65535 (X= 1, Y ranges from 1 to 12) Description: Configures the priority for the target host in SRV record Y.
  • Page 261 None account.X.dns_cache_a.Y.name Domain name Blank (X= 1, Y ranges from 1 to 12) Description: Configures the domain name in A record Y. Example: account.1.dns_cache_a.1.name = yealink.pbx.com Web User Interface: None Phone User Interface: None account.X.dns_cache_a.Y.ip IP address Blank (X= 1, Y ranges from 1 to 12) Description: Configures the IP address that the domain name in A record Y maps to.
  • Page 262 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default account.X.dns_cache_a.Y.ttl Integer from 30 to 2147483647 (X= 1, Y ranges from 1 to 12) Description: Configures the time interval (in seconds) that A record Y may be cached before the record should be consulted again.
  • Page 263: Lldp

    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.
  • Page 264 Administrator’s Guide for CP860 IP conference phones TLV Type TLV Name Description End of LLDPDU Marks end of LLDPDU. Name assigned to the IP phone. System Name The default value is “yealink”. Description of the IP phone. System Description The default value is “yealink”.
  • Page 265 Inventory – Serial Serial number of phone. Number Inventory – Manufacturer name of phone. Manufacturer Name The default value is “yealink”. Inventory – Model Model name of phone. Name Assertion identifier of phone. Asset ID The default value is “asset”.
  • Page 266 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Phone User Interface: None network.lldp.packet_interval Integer from 1 to 3600 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.
  • Page 267: Vlan

    DHCP , the IP phone will examine DHCP option for a valid VLAN ID. The predefined option 132 is used to supply the VLAN ID by default. You can customize the DHCP option used to request the VLAN ID. VLAN Feature on Yealink IP Phones For more information on VLAN, refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 268 Administrator’s Guide for CP860 IP conference phones feature. Navigate to: http://<phoneIPAddress>/servlet? p=network-adv&q=load Configure VLAN for the Internet Phone User Interface port. Details of Configuration Parameters: Parameters Permitted Values Default network.vlan.internet_port_enable 0 or 1 Description: Enables or disables VLAN for the Internet (WAN) port.
  • Page 269 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 Description:...
  • Page 270 Administrator’s Guide for CP860 IP conference phones Select the desired value (0-7) from the pull-down list of Priority. Click Confirm to accept the change. A dialog box pops up to prompt reboot to make the settings effective. Click OK to reboot the phone.
  • Page 271: Vpn

    Configuring Advanced Features The default option is 132. Click Confirm to accept the change. A dialog box pops up to prompt that settings will take effect after reboot. Click OK to reboot the phone. To configure VLAN for Internet port via phone user interface: Press Menu->Settings->Advanced Settings (Default password: admin) ->Network->VLAN->WAN Port.
  • Page 272 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.
  • Page 273 Configuring Advanced Features Details of Configuration Parameters: Parameters Permitted Values Default network.vpn_enable 0 or 1 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.
  • Page 274: Quality Of Service

    Administrator’s Guide for CP860 IP conference phones Click Upload to upload the TAR file. The web user interface prompts the message “Import config…”. In the VPN block, select the desired value from the pull-down list of Active. Click Confirm to accept the change.
  • Page 275 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 ...
  • Page 276 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.
  • Page 277 Configuring Advanced Features Parameters Permitted Values Default None network.qos.signaltos Integer from 0 to 63 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.
  • Page 278: Network Address Translation

    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.
  • Page 279 Configuring Advanced Features Details of Configuration Parameters: Parameters Permitted Values Default account.X.nat.nat_traversal 0 or 1 (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 IP address or domain name Blank (X = 1) Description:...
  • Page 280: Snmp

    Administrator’s Guide for CP860 IP conference phones To configure the NAT traversal and STUN server via web user interface: Click on Account. Select STUN from the pull-down list of NAT. Enter the IP address or the domain name in the STUN Server field.
  • Page 281 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.
  • Page 282 Administrator’s Guide for CP860 IP conference phones Configure SNMP . Navigate to: Local Web User Interface http://<phoneIPAddress>/servl et?p=network-adv&q=load Details of Configuration Parameters: Parameters Permitted Values Default network.snmp.enable 0 or 1 Description: Enables or disables SNMP feature on the IP phone.
  • Page 283 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...
  • Page 284: X Authentication

    Administrator’s Guide for CP860 IP conference phones Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. 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.
  • Page 285 Configuring Advanced Features Details of Configuration Parameters: Parameters Permitted Values Default network.802_1x.mode 0, 1, 2, 3 or 4 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.
  • Page 286 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...
  • Page 287 Configuring Advanced Features To configure the 802.1X via web user interface: Click on Network->Advanced. In the 802.1x block, select the desired protocol from the pull-down list of 802.1x Mode. a) If you select EAP-MD5: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field.
  • Page 288 Administrator’s Guide for CP860 IP conference phones 5) Click Upload to upload the certificates. c) If you select PEAP-MSCHAPv2: 1) Enter the user name for authentication in the Identity field. 2) Enter the password for authentication in the MD5 Password field.
  • Page 289 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.
  • Page 290 Administrator’s Guide for CP860 IP conference phones 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. Click Save to accept the change.
  • Page 291 AddObject object defined on the CPE. This method is used to remove a particular instance DeleteObject of an object. Yealink TR-069 Technote For more information on TR-069, refer to , available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142. Procedure TR-069 can be configured using the configuration files or locally.
  • Page 292 Administrator’s Guide for CP860 IP conference phones managementserver.periodic_inform_interval Configure the TR-069 feature. Web User Navigate to: Local Interface http://<phoneIPAddress>/servlet?p=settings-prefer ence&q=load Details of Configuration Parameters: Permitted Parameters Default Values managementserver.enable 0 or 1 Description: Enables or disables TR-069 feature. 0-Disabled 1-Enabled Web User Interface: Settings->TR069->Enable TR069...
  • Page 293 Configuring Advanced Features Permitted Parameters Default Values required. Example: managementserver.password = pwd123 Web User Interface: Settings->TR069->ACS Password Phone User Interface: None URL within 511 managementserver.url Blank characters 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:...
  • Page 294 Administrator’s Guide for CP860 IP conference phones Permitted Parameters Default Values 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.periodic_inform_enable 0 or 1...
  • Page 295 Configuring Advanced Features Enter the URL of the ACS in the ACS URL field. Select the desired value from the pull-down list of Enable Periodic Inform. Enter the desired time in the Periodic Inform Interval (seconds) field. Enter the user name and password authenticated by the IP phone in the Connection Request Username and Connection Request Password fields.
  • Page 296 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 network.ipv6_internet_port.ip Configuration File <MAC>.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.
  • Page 297 Configuring Advanced Features Parameters Permitted Values Default Port->IP Mode network.ipv6_internet_port.type 0 or 1 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.
  • Page 298 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.ipv6_prefix Integer from 0 to 128 Description: Configures the IPv6 prefix 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...
  • Page 299 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.
  • Page 300 Administrator’s Guide for CP860 IP conference phones 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.
  • Page 301 Configuring Advanced Features A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the phone. To configure IPv6 address via phone user interface: Press Menu->Settings->Advanced Settings (Default password: admin) ->Network->WAN Port. Press the soft key to select the desired address mode from the IP Mode field.
  • Page 302 Administrator’s Guide for CP860 IP conference phones...
  • Page 303: Configuring Audio Features

    This can effectively reduce the frame size and the bandwidth required for audio transmission. The following table lists the audio codecs supported by CP860 IP conference phones: Supported Audio Codecs Default Audio Codecs...
  • Page 304 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.
  • Page 305 Configuring Audio Features Codec Configuration Methods Priority RTPmap Configuration Files iLBC Web User Interface 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: account.X.codec.Y.priority Configuration File...
  • Page 306 Administrator’s Guide for CP860 IP conference phones Parameters Permitted Values Default Description: Enables or disables the specified codec. 0-Disabled 1-Enabled When Y=1, the default value is 1; When Y=2, the default value is 1; When Y=3, the default value is 0;...
  • Page 307 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 Refer to the Integer from 0 to 11...
  • Page 308 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;...
  • Page 309 Configuring Audio Features To configure the codecs and adjust the priority of the enabled codecs on a per-line basis via web user interface: Click on Account->Codec. Select the desired codec from the Disable Codecs column and click The selected codec appears in the Enable Codecs column. Repeat the step 2 to add more codecs to the Enable Codecs column.
  • Page 310: Acoustic Echo Cancellation

    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.
  • Page 311: Background Noise Suppression

    Configuring Audio Features Select the desired value from the pull-down list of ECHO. 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.
  • Page 312 Administrator’s Guide for CP860 IP conference phones voice.vad Configure VAD. Navigate to: Local Web User Interface http://<phoneIPAddress>/servl et?p=settings-voice&q=load Details of the Configuration Parameter: Parameter Permitted Values Default voice.vad 0 or 1 Description: Enables or disables VAD (Voice Activity Detection) feature on the IP phone.
  • Page 313: Comfort Noise Generation

    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.
  • Page 314: Jitter Buffer

    Administrator’s Guide for CP860 IP conference phones Select the desired value from the pull-down list of CNG. 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.
  • Page 315 Configuring Audio Features jitter buffer. Navigate to: http://<phoneIPAddress>/servl et?p=settings-voice&q=load Details of Configuration Parameters: Parameters Permitted Values Default voice.jib.adaptive 0 or 1 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 Integer from 0 to 400 Description: Configures the minimum delay time (in milliseconds) of jitter buffer.
  • Page 316 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...
  • Page 317: Configuring Security Features

    CP860 IP conference phones support TLS 1.0. A cipher suite is a named combination of authentication, encryption, and message authentication code (MAC) algorithms used to negotiate the security settings for a network connection using the TLS/SSL network protocol.
  • Page 318 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 ...
  • Page 319 *.pem and *.cer. A unique server certificate: It is installed by default and is unique to an IP phone (based on the MAC address) and issued by the Yealink Certificate Authority (CA). A generic server certificate: It is installed by default and is issued by the Yealink Certificate Authority (CA).
  • Page 320 Administrator’s Guide for CP860 IP conference phones Procedure Configuration changes can be performed using the configuration files or locally. Configure TLS. <MAC>.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.
  • Page 321 Configuring Security Features et?p=server-cert&q=load Details of Configuration Parameters: Parameters Permitted Values Default account.X.transport Integer (X = 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 0 or 1 Description: Enables or disables the IP phone to only trust the server certificates in the Trusted Certificates list.
  • Page 322 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...
  • Page 323 Configuring Security Features Parameters Permitted Values Default URL within 511 trusted_certificates.url Blank characters 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.
  • Page 324 Administrator’s Guide for CP860 IP conference phones Select the desired value from the pull-down list of CA Certificates. Click Confirm to accept the change. A dialog box pops up to prompt that the settings will take effect after reboot. Click OK to reboot the phone.
  • Page 325 Configuring Security Features Click Confirm to accept the change. To upload a trusted certificate via web user interface: Click on Security->Trusted Certificates. Click Browse to locate the certificate (*.pem,*.crt, *.cer or *.der) from your local system. Click Upload to upload the certificate. To configure the server certificates feature via web user interface: Click on Security->Server Certificates.
  • Page 326: Secure Real-Time Transport Protocol

    Administrator’s Guide for CP860 IP conference phones Click Browse to locate the certificate (*.pem or *.cer) from your local system. Click Upload to upload the certificate. The dialog box pops up to prompt “Success: The Server Certificate has been loaded! Rebooting, please wait…”.
  • Page 327 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. Example of the RTP encryption algorithm carried in the SDP of the 200 OK message: m=audio 11780 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000...
  • Page 328: Encrypting Configuration Files

    Encrypted configuration files can be downloaded from the provisioning server to protect against unauthorized access and tampering of sensitive information (e.g., login passwords, registration information). Yealink supplies a configuration encryption tool for encrypting configuration files. The encryption tool encrypts plaintext y000000000037.cfg...
  • Page 329: Procedure To Encrypt Configuration Files

    This tool generates another new file named as Aeskey.txt to store the plaintext 16-character symmetric keys for each configuration file. For a Microsoft Windows platform, you can use a Yealink-supplied encryption tool "Config_Encrypt_Tool.exe" to encrypt the y000000000037.cfg and <MAC>.cfg files respectively.
  • Page 330 Administrator’s Guide for CP860 IP conference phones automatically in the directory where the application tool is located. 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.
  • Page 331 Configuring Security Features 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.
  • Page 332 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 <MAC_Security>.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).
  • Page 333 Configuring Security Features Parameters Permitted Values Default Phone User Interface: None auto_provision.update_file_mode 0 or 1 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: Click on Settings->Auto Provision.
  • Page 334 Administrator’s Guide for CP860 IP conference phones...
  • Page 335: Resource Files

    However, if you want to specify the desired phone to use the resource file, the access URL of resource file should be specified in the <MAC>.cfg file. The names of the Yealink-supplied template file are (You can rename the filename as required):...
  • Page 336 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.
  • Page 337 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: <DialNow>...
  • Page 338 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.
  • Page 339 Resource Files <Key Type="Switch"/> <Key Type="Cancel"/> </Disable> <Enable> <Key Type="NewCall"/> <Key Type="Empty"/> <Key Type="Empty"/> <Key Type="Empty"/> </Enable> <Default> <Key Type="NewCall"/> <Key Type="Empty"/> <Key Type="Empty"/> <Key Type="Empty"/> </Default> </CallFailed> Directory provides easy access to frequently used lists. Users can access lists by pressing the Directory soft key when the IP phone is idle.
  • Page 340 Administrator’s Guide for CP860 IP conference phones Procedure Use the following procedures to customize a directory template. Customizing a directory template: Open the template file using an ASCII editor. For each directory list that you want to configure, edit the corresponding string in the file.
  • Page 341 Resource Files When editing a super search template, learn the following: <root_super_search> indicates the start of a template and </root_super_search>  indicates the end of a template. The default display names of directory lists are Local Directory, History, Remote  Phone Book and LDAP .
  • Page 342 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.
  • Page 343 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. Specify the values within double quotes. Save the change and place this file to the provisioning server.
  • Page 344 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: Open the template file using an ASCII editor. For each department that you want to add, add the following strings to the file.
  • Page 345 Resource Files <Name>#</Name> <URL>http://10.3.6.117:8080/TextMenu</URL> </SoftKeyItem> </YealinkIPPhoneMenu> When creating a Department.xml file, learn the following: <YealinkIPPhoneDirectory> indicates the start of a department file and  </YealinkIPPhoneDirectory> indicates the end of a department file. Create contact lists for a department between <DirectoryEntry> and ...
  • Page 346 </DirectoryEntry> <DirectoryEntry> <Name>John</Name> <Telephone>1004</Telephone> </DirectoryEntry> <DirectoryEntry> <Name>Marry</Name> <Telephone>1005</Telephone> </DirectoryEntry> </YealinkIPPhoneDirectory> Note Yealink supplies a phone book generation tool to quickly generate a remote XML phone Yealink Phonebook Generation Tool User Guide book. For more information, refer to available online: http://www.yealink.com/DocumentDownload.aspx?CateId=142&flag=142.
  • Page 347: Troubleshooting

    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.
  • Page 348 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 export the log files.
  • Page 349 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:...
  • Page 350 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: Click on Settings->Configuration. Select 6 from the pull-down list of System Log Level.
  • Page 351 Troubleshooting Enter the IP address or domain name of the syslog server in the Server Name field. 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. Click OK to reboot the phone.
  • Page 352 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.
  • Page 353 Troubleshooting 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.
  • Page 354 Administrator’s Guide for CP860 IP conference phones Details of the Configuration Parameter: Parameter Permitted Values Default watch_dog.enable 0 or 1 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.
  • Page 355 This section describes solutions to common issues that may occur while using the IP phone. Upon encountering a scenario not listed in this section, contact your Yealink reseller for further support.
  • Page 356 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 ...
  • Page 357 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 ...
  • Page 358 Administrator’s Guide for CP860 IP conference phones From: sip:sipsak@<srchost> CSeq: 10 NOTIFY Call-ID: 1234@<srchost> 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.
  • Page 359 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. ...
  • Page 360 Administrator’s Guide for CP860 IP conference phones The web user interface prompts the message “Do you want to reset to factory?”. 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.
  • Page 361: Appendix

    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).
  • Page 362 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.
  • Page 363 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...
  • Page 364 Administrator’s Guide for CP860 IP conference phones Time Zone Time Zone Name United Kingdom(London) Morocco +01:00 Albania(Tirane) +01:00 Austria(Vienna) +01:00 Belgium(Brussels) +01:00 Caicos +01:00 Chad +01:00 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)
  • Page 365 Tonga(Nukualofa) This appendix describes the programable key parameters you can configure on IP phones. Programable keys can be assigned with various key features. The CP860 IP phones support 8 programmble keys. The programable key takes effect only if the IP phone is idle.
  • Page 366 Administrator’s Guide for CP860 IP conference phones Valid types are:  Forward   Call Return  Intercom  XML Group  Multicast Paging  History  Menu  Status  LDAP  Prefix  Local Directory  Local Group ...
  • Page 367 Appendix 45-Local Group Directory 47-XML 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 parameter is used to add a specified prefix Example number before the dialed number.
  • Page 368 Configures the second remote phone book. Example programablekey.1.xml_phonebook = 1 This section describes how Yealink CP860 IP conference phones comply with the IETF definition of SIP as described in RFC 3261. This section contains compliance information in the following: RFC and Internet Draft Support ...
  • Page 369 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 ...
  • Page 370 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 ...
  • Page 371: Sip Request

    To find the applicable Request for Comments (RFC) document, go to http://www.ietf.org/rfc.html and enter the RFC number. The following SIP request messages are supported: Method Supported Notes REGISTER Yealink CP860 IP conference phones INVITE support mid-call changes such as putting a call on hold as signaled by a new...
  • Page 372 Administrator’s Guide for CP860 IP conference phones Method Supported Notes INVITE that contains an existing Call-ID. CANCEL OPTIONS SUBSCRIBE NOTIFY REFER PRACK INFO MESSAGE UPDATE PUBLISH The following SIP request headers are supported: Method Supported Notes Accept Alert-Info Allow Allow-Events...
  • Page 373: Sip Responses

    Appendix Method Supported Notes Event Expires From Max-Forwards Min-SE P-Asserted-Identity P-Preferred-Identity Proxy-Authenticate Proxy-Authorization RAck Record-Route Refer-To Referred-By Remote-Party-ID Replaces Require Route RSeq Session-Expires Subscription-State Supported User-Agent The following SIP responses are supported: 1xx Response—Information Responses 1xx Response Supported Notes...
  • Page 374 Administrator’s Guide for CP860 IP conference phones 1xx Response Supported Notes 100 Trying 180 Ringing 181 Call Is Being Forwarded 183 Session Progress 2xx Response—Successful Responses 2xx Response Supported Notes 200 OK 202 Accepted In REFER transfer. 3xx Response—Redirection Responses...
  • Page 375 Appendix 4xx Response Supported Notes 411 Length Required 413 Request Entity Too Large 414 Request-URI Too Long 415 Unsupported Media Type 416 Unsupported URI Scheme 420 Bad Extension 421 Extension Required 423 Interval Too Brief 480 Temporarily Unavailable 481 Call/Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops...
  • Page 376 Administrator’s Guide for CP860 IP conference phones 6xx Response—Global Responses 6xx Response Supported Notes 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable SDP Headers Supported v—Protocol version o—Owner/creator and session identifier a—Media attribute c—Connection information m—Media name and transport...
  • Page 377  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: User A calls User B.
  • Page 378 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 ...
  • Page 379 The following figure illustrates the scenario of an unsuccessful call caused by the called user’s being busy. 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.
  • Page 380 Administrator’s Guide for CP860 IP conference phones The call flow scenario is as follows: User A calls User B. User B is busy on the IP phone and unable or unwilling to take another call. The call cannot be set up successfully.
  • Page 381 Appendix Step Action Description is specified. The proxy server maps the SIP URI in the INVITE—Proxy Server to User To field to User B. Proxy server forwards the INVITE message to User B. User B sends a SIP 100 Trying response to the proxy server.
  • Page 382 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: User A calls User B.
  • Page 383 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.
  • Page 384: Successful Call Setup And Call Hold

    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.
  • Page 385 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 ...
  • Page 386 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.
  • Page 387 Appendix The call flow scenario is as follows: User A calls User B. User B answers the call. User C calls User B. User B accepts the call from User C. Proxy Server User C User A User B F1. INVITE B F2.
  • Page 388 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 ...
  • Page 389 Appendix Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to User B.
  • Page 390 Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends a mid-call INVITE request INVITE—User A to Proxy to the proxy server with new SDP Server session parameters, which are used to place the call on hold.
  • Page 391 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.
  • Page 392 Administrator’s Guide for CP860 IP conference phones 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.
  • Page 393 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 ...
  • Page 394 Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active.
  • Page 395 This is called attended transfer. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
  • Page 396 Administrator’s Guide for CP860 IP conference phones User A transfers the call to User C. Call is established between User B and User C. User A Proxy Server User B User C F1. INVITE B F2. INVITE B F3. 180 Ringing F4.
  • Page 397 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 ...
  • Page 398 Administrator’s Guide for CP860 IP conference phones Step Action Description User A sends a SIP ACK to the proxy server, The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active.
  • Page 399 Appendix Step Action Description sends the INVITE request to User C. User C sends a SIP 180 Ringing 180 Ringing—User C to Proxy response to the proxy server. The 180 Server Ringing response indicates that the user is being alerted. The proxy server forwards the 180 180 Ringing—Proxy Server to Ringing response to User A.
  • Page 400: Always Call Forward

    User C when User A calls User B. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network.
  • Page 401 Appendix 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.
  • Page 402 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 ...
  • Page 403 Appendix Step Action Description User A sends a SIP INVITE request to the proxy server. In the INVITE request, a INVITE—User A to Proxy unique Call-ID is generated and the Server Contact-URI field indicates that User A requested the call. The proxy server maps the SIP URI in the INVITE—Proxy Server to User To field to User C.
  • Page 404: Busy Call Forward

    User B is busy. In this call flow scenario, the end users are User A, User B, and User C. They are all using Yealink SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: User B enables busy call forward, and the destination number is User C.
  • Page 405 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 ...
  • Page 406 Administrator’s Guide for CP860 IP conference phones Step Action Description ACK message. 302 Move Temporarily—Proxy The proxy server forwards the 302 Server to User A Moved Temporarily message to User A. User A sends a SIP ACK to the proxy server.
  • Page 407: No Answer Call Forward

    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.
  • Page 408 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 ...
  • Page 409 Appendix Step Action Description ACK message. 302 Move Temporarily—Proxy The proxy server forwards the 302 Server to User A Moved Temporarily message to User A. User A sends a SIP ACK to the proxy server. The ACK message notifies the ACK—User A to Proxy Server proxy server that User A has received the ACK message.
  • Page 410 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.
  • Page 411 Appendix User A mixes the RTP channels and establishes a conference between User B and User C. User A User B User C Proxy Server F1. INVITE B F2. INVITE B F3. 180 Ringing F4. 180 Ringing F5. 200 OK F6.
  • Page 412 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 ...
  • Page 413 Appendix Step Action Description User A sends a SIP ACK to the proxy server. The ACK confirms that User A ACK—User A to Proxy Server has received the 200 OK response. The call session is now active. The proxy server sends the SIP ACK to User B.
  • Page 414 Administrator’s Guide for CP860 IP conference phones Step Action Description sends the SIP INVITE request to User C. User C sends a SIP 180 Ringing 180 Ringing—User C to Proxy response to the proxy server. The 180 Server Ringing response indicates that the user is being alerted.
  • Page 415 Index Numeric Call Log Call Return 180 Ring Workaround Call Transfer 802.1x Authentication Call Waiting Call Waiting Tone Calling Line Identification Presentation About This Guide Connected Line Identification Presentation Acoustic Echo Cancellation Capturing Packets Action URL Comfort Noise Generation Action URI Configuration Files Administrator Password Configuration Methods...
  • Page 416 Administrator’s Guide for CP860 IP conference phones H.323 Phone Lock Hotline Phone User Interface Physical Features of CP860 IP conference phones Product Overview In This Guide Index Initialization Process Overview Intercom Quality of Service IPv6 Support Reading Icons Jitter Buffer...
  • Page 417: Troubleshooting Solutions

    Index Transfer on Conference Hang Up Transfer via DTMF Transport Layer Security (TLS) Troubleshooting Troubleshooting Methods Troubleshooting Solutions TR-069 Device Management Upgrading Firmware Use Outbound Proxy in Dialog User Agent Client (UAC) User Agent Server (UAS) User Password Verifying Startup Viewing Log Files VLAN Voice Activity Detection...

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