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Cisco 8831 Administration Manual

Unified ip conference phone for third-party call control
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Cisco Unified IP Conference Phone 8831
for Third-Party Call Control Administration
Guide, Release 9.3(4)
December 4, 2014
Revised: October 10, 2016
Cisco Systems, Inc.
www.cisco.com
Cisco has more than 200 offices worldwide.
Addresses, phone numbers, and fax numbers
are listed on the Cisco website at
www.cisco.com/go/offices.
ADMINISTRATION GUIDE

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  Summary of Contents for Cisco 8831

  • Page 1 Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide, Release 9.3(4) December 4, 2014 Revised: October 10, 2016 Cisco Systems, Inc. www.cisco.com Cisco has more than 200 offices worldwide. Addresses, phone numbers, and fax numbers are listed on the Cisco website at www.cisco.com/go/offices.
  • Page 2 OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. To view a list of Cisco trademarks, go to this URL: www.cisco.com/go/trademarks.
  • Page 3 Configure the Phone Name Customize the Startup Screen Change the Display Background Picture Configure the Screen Saver Configure the LCD Contrast Configure Back Light Settings Call Appearances Per Line Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 4: Table Of Contents

    Enable and Configure the Phone Web Server Configure the Web Server from the Phone Web Interface Configure the Web Server from the Phone Screen Interface Configure LDAP for the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Configure BroadSoft Settings 2-10...
  • Page 5 Dial Plan Timer (Off-Hook Timer) Syntax for the Dial Plan Timer Interdigit Long Timer (Incomplete Entry Timer) Syntax for the Interdigit Long Timer Interdigit Short Timer (Complete Entry Timer) Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 6 Configure Daylight Saving Time Daylight Saving Time Examples Select a Display Language Create a Dictionary Server Script Localization Configuration Example Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Info System Status System Information Reboot History Product Information...
  • Page 7 A-24 Call Feature Settings A-24 Call History A-24 Related Documentation Cisco IP Phone 8800 Series Documentation Cisco IP Phone Firmware Support Policy Documentation, Service Requests, and Additional Information Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 8 Contents Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide viii...
  • Page 9 About This Document This guide describes the administration of the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control: Purpose, page ix • Document Audience, page ix • Organization, page ix • Document Conventions, page x • Purpose The document describes the administration of Cisco Unified IP Conference Phone 8831 for Third-Party Call Control devices.
  • Page 10 Angle brackets (<>) identify parameters that appear on the configuration pages of the administration web server. Italic A variable that should be replaced with a literal value. A code sample or system output. Monospaced Font Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 11 For more information on phone features, see the data sheets for this product. Network Configurations The Cisco Unified IP Conference Phone 8831 for Third-Party Call Control is used as a part of a SIP network as it supports Session Initiation Protocol (SIP).
  • Page 12 Click Undo All Changes if you want to clear all changes made this session and return to the parameter values set before the session began or since the last time you clicked Submit All Changes. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 13 User account name is user. These account names cannot be changed. The Admin account is designed to give the service provider or VAR configuration access to the Cisco IP phone, while the User account is designed to give limited and configurable control to the end user of the device.
  • Page 14 View Phone Information You can check the current status of the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control by clicking the Info tab. The Info tab shows information about all phone extensions, including phone statistics and the registration status.
  • Page 15 For all devices, no matter the firmware release, no wireless region setting is available to the user. If your Cisco Unified IP Conference Phone 8831 for Third-Party Call Control is operating with an earlier firmware release, you must upgrade to firmware release 9.3(4) so that the Wireless Microphone Region can be locked.
  • Page 16 To do so, execute Info > System Status > Product Information and check the Wireless Microphone Region value on the webpage. Refer to the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Release Notes for Firmware Release 9.3(4) if you need to upgrade a device from firmware release 9.3(3) to 9.3(4).
  • Page 17 You can create a text or 128-by-48 pixel by 1-bit deep image logo to display when the conference phone boots up. A logo displays during the boot sequence for a short period after the Cisco logo displays. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 18 Click Admin Login > advanced > Voice > Phone. Step 3 Select the background picture in the Select Background Picture menu: • None–Does not display a background picture. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 19 • “Press any key to unlock your phone.” Cisco logo. • The station date and time on the IP phone screen. • Click Submit All Changes. Step 4 Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 20: Enable Call Features

    Enable Call Transfer and Call Forwarding Services You can transfer or forward a call when the service is enabled. Click Admin Login > advanced > Voice > Phone. Step 1 Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 21: Enable Conferencing

    You can allow users to turn the Do Not Disturb feature on or off. This feature plays a message to the caller saying the user is unavailable. On the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control, the users can press the Ignore softkey to divert a ringing call to another destination.
  • Page 22: Configure Audio Settings

    Administrators and users have different privileges and see different options for the phone based on their role. Configure the Web Server from the Phone Web Interface To enable the web server: Click Admin Login > advanced > System. Step 1 Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 23: Configure The Web Server From The Phone Screen Interface

    Chapter 2 Customize Standard Features Configure LDAP for the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Under the System Configuration section in the Enable Web Server field, verify that the parameter is Step 2 set to Yes to enable the web administration server.
  • Page 24 Chapter 2 Customize Standard Features Configure LDAP for the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control In the Optional Network Configuration section, under Domain, enter the LDAP domain. (Only Step 3 required if using Active Directory with authentication set to MD5.) Some sites might not deploy DNS internally and instead use Active Directory 2003.
  • Page 25 Chapter 2 Customize Standard Features Configure LDAP for the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Parameter Description LDAP First Name Filter Define the search for the common name . For example, [cn] . This searches for the text string anywhere in the cn:(cn=*$VALUE*) beginning, middle, or at the end of a name.
  • Page 26: Configure Broadsoft Settings

    Dir. Name Admin password required (if set) Host Server Admin password required (if set) Type None User ID None Password None Click Submit All Changes. Step 3 Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide 2-10...
  • Page 27: Configure Sip And Nat

    C H A P T E R Configure SIP and NAT The Cisco Unified IP Conference Phone 8831 for Third-Party Call Control uses the following protocol: Session Initiation Protocol (SIP) • This chapter describes how to configure the SIP phone protocol: •...
  • Page 28: Sip Over Tcp

    Cisco IP phone. A simple way to support proxy redundancy is to configure a SIP proxy server in the Cisco conference phone configuration profile. The DNS SRV records instruct the phones to contact a SIP proxy server in a domain named in SIP messages.
  • Page 29: Limitations For Dual Registration And Dns Srv Redundancy

    800 seconds. After successfully registering back to primary server, all the SIP messages go to primary server. RFC3261 Support The Cisco Unified IP Conference Phone 8831 for Third-Party Call Control supports RFC-3261, the SIP UPDATE Method. Support for SIP NOTIFY XML-Service The Cisco Unified IP Conference Phone 8831 for Third-Party Call Control support the SIP NOTIFY XML-Service event.
  • Page 30: Configure Sip

    A1 = username ":" realm ":" passwd and A2 = Method ":" digest-uri Configure SIP SIP settings for the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control are configured for the phone in general and for the extensions. Configure Basic SIP Parameters To configure general SIP parameters, navigate to Admin Login >...
  • Page 31: Configure Sip Timer Values

    If this interval is 0, the phone stops trying. This value should be much larger than the Reg Retry Intvl value. The range is from 0 to 2147483647. Defaults to 1200 seconds. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 32: Configure Response Status Code Handling

    RTP Packet Size—Packet size in seconds. The range is from 0.01 to 0.16. Valid values must be a • multiple of 0.01 seconds. Defaults to 0.02. RTCP Tx Enable—To enable Real-Time Transport Control Protocol (RTCP) sender report on an • active connection. Defaults to no. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 33: Configure Sdp Payload Types

    Configure SIP and NAT Configure SIP During an active connection, the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control sends out compound RTCP packets. Each compound RTP packet, except the last one, contains a sender report (SR) and a source description (SDES). The last RTCP packet contains an additional BYE packet.
  • Page 34: Configure A Sip Proxy Server

    The port number is optional. The default is port 5060. Outbound Proxy All outbound requests are sent as the first hop. Enter an IP address or domain name. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 35 The range is from 32 to 2000000. Defaults to 3600 seconds. Use DNS SRV Enables DNS SRV lookup for the proxy and outbound proxy. To enable this feature, select Yes. Otherwise, select No. Defaults to No. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 36: Configure Subscriber Information Parameters

    Enter the public IP address for your router. Step 3 Click the Extension tab and navigate to NAT Settings. Step 4 Set NAT Keep Alive Enable to Yes. Step 5 Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide 3-10...
  • Page 37 Step 6 Click Submit All Changes. Configure the firewall settings on your router to allow SIP traffic. See the “Configure SIP” section on Step 7 page 3-4. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide 3-11...
  • Page 38 Chapter 3 Configure SIP and NAT Configure NAT Support Parameters Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide 3-12...
  • Page 39: Configure Security, Quality, And Network Features

    Configure Domain and Internet Settings Configure Restricted Access Domains If you enter domains, the Cisco IP phones respond to SIP messages only from the identified servers. To configure restricted access domains, navigate to Admin Login > advanced > Voice > System. Under System Configuration in the Restricted Access Domains field.
  • Page 40: Challenge Sip Initial Invite Messages

    SIP servers that are permitted to interact with the devices on a service provider network. This significantly increases the security of the VoIP network by preventing malicious attacks against the device. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 41: Encrypt Signaling With Sip Over Tls

    Note that the conference phone supports voice codec priority. You can select up to three preferred codecs. The administrator can select the low-bit-rate codec used for each line. G.711a and G.711u are always enabled. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 42 AVT sends DTMF as AVT events. • INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec • negotiation. Defaults to Auto. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 43: Set Optional Network Servers

    Domain—The network domain of the phone. If using LDAP see the “Configure LDAP for the Cisco • Unified IP Conference Phone 8831 for Third-Party Call Control” section on page 2-7. • Primary DNS—DNS server used by the phone in addition to the DHCP-supplied DNS servers (if DHCP is enabled), When DHCP is disabled, this is the primary DNS server.
  • Page 44: Configure Cisco Discovery Protocol (Cdp)

    Configure VLAN Settings Configure Cisco Discovery Protocol (CDP) CDP is negotiation-based and determines which VLAN the IP phone resides in. If you are using a Cisco switch, Cisco discovery protocol (CDP) is available and is enabled by default. CDP: •...
  • Page 45: Tlv Information

    Time to live TLV • End of LLDPDU TLV • There are some restrictions in the implementation of LLDP-MED on the Cisco IP Phones: Storage and retrieval of neighbor information is not supported. • SNMP and corresponding MIBs are not supported.
  • Page 46 Configure Security, Quality, and Network Features Configure VLAN Settings System Name TLV For the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control, the value is SEP+MAC address. Example: SEPAC44F211B1D0 The incoming LLDPDU, the System Name TLV, is ignored and not validated. Only one System Name TLV is allowed for the outgoing and incoming LLDPDUs.
  • Page 47 ID valid value is range from 1-4094. However, VLAN ID=1 will never be used (limitation). If DSCP is used, the value range from 0-63 is set accordingly. Incoming LLDPDU—Multiple Network Policy TLVs for different application types are allowed. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 48 If the VLAN > 1 and VLAN< 4095, the VLAN is set accordingly. CoS and ToS are based on the • default as previously described. DSCP is applicable. The phone reboots and restarts the fast start sequence. • Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide 4-10...
  • Page 49 The phones do not support IEEE 802.X and will not work in a 802.1X wired environment. However, IEEE 802.1X or Spanning Tree Protocols on network devices could result in delay of fast start response from switches. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide 4-11...
  • Page 50: Configure The Vlan Settings

    LLDP-MED to work. Configuring a delay can be important for networks that use Spanning Tree Protocol. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide 4-12...
  • Page 51: Provisioning

    The use of a FQDN facilitates the deployment of redundant provisioning servers. When the provisioning server is identified through a FQDN, the Cisco IP phone attempts to resolve the FQDN to an IP address through DNS. Only DNS A-records are supported for provisioning; DNS SRV address resolution is not available for provisioning.
  • Page 52: Retail Provisioning

    Upon receiving the unit, the customer connects the unit to the broadband link. On power-up the Cisco IP phone already knows the server to contact for its periodic resync update.
  • Page 53: Use Https

    Generate a private key that will be used in a CSR (Certificate Signing Request). This key is private and Step 2 you do not need to provide this key to Cisco support. Use open source “openssl” to generate the key. For example: openssl genrsa -out <file.key>...
  • Page 54: Manually Provision A Phone From The Keypad

    HTTP, or 443 for HTTPS). Step 3 Press the Resync softkey. Sample Configuration File Refer to the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Provisioning Guide. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 55: Update Profiles And Firmware

    A profile resync is only attempted when the Cisco IP phone is idle, because this might trigger a software reboot and disconnect a call. General purpose parameters manage the provisioning process. Each Cisco IP phone can be configured to periodically contact a normal provisioning server (NPS).
  • Page 56 The syslog message issued upon successful completion of a resync attempt. The default value is: $PN $MAC –Successful % $SCHEME://$SERVIP:$PORT$PATH -- $ERR User Configurable Resync Allows a user to resync the phone from the phone screen. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 57: Allow And Configure Firmware Updates

    A device (with new base and DCU) may not be downgraded to an earlier firmware release, such as 9.3(3). For details, refer to the hardware information and the firmware/hardware compatibility information in the current Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Release Notes. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 58: Firmware Upgrade With A Browser Command

    An upgrade command entered into the browser address bar can be used to upgrade firmware on a phone. The phone updates only when it is idle. The update is attempted automatically after the call is complete. To upgrade the conference phone CP-8831-3PCC via URL on web browser enter this command: http://<phone_ip>/admin/upgrade?<schema>://<serv_ip[:port]>/filepath Configure a Custom Certificate Authority Digital certificates can be used to authenticate network devices and users on the network.
  • Page 59: General Purpose Parameters

    These parameters can be used as variables in provisioning and upgrade rules. They are referenced by prepending the variable name with a ‘$’ character, such as $GPP_A. To configure general purpose parameters, navigate to Admin Login > advanced > Voice > Provisioning. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 60 Chapter 5 Provisioning General Purpose Parameters Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide 5-10...
  • Page 61: Configure Dial Plan

    0 1 2 3 4 5 6 7 8 9 0 * # Characters that represent a key that the user must press on the phone keypad. Any character on the phone keypad. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 62 0 delay for the hot line and a non-zero delay for a warm line. For example: EXAMPLE: P5 A pause of 5 seconds is introduced. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 63: Digit Sequence Examples

    1-900 numbers in the U.S.. After the user press 9, an external dial tone sounds. If the user enters an 11-digit number that starts with the digits 1900, the call is rejected. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 64: Acceptance And Transmission Of The Dialed Digits

    If the sequence is incomplete or is blocked by the dial plan, the number is rejected. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 65: Dial Plan Timer (Off-Hook Timer)

    This section explains how to edit a timer as part of a dial plan. Alternatively, you can modify the Control Timer that controls the default interdigit timers for all calls. See the “Reset the Control Timers” section on page 6-7. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 66: Syntax For The Interdigit Long Timer

    EXAMPLE: (9,8<:1408>[2-9]xxxxxx | | 9,8,011xx. | 9,8,xx.|[1-8]xx) 9,8,1[2-9]xxxxxxxxxS0 With the timer set to 0, the call is transmitted automatically when the user dials the final digit in the sequence. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 67: Edit Dial Plan On The Ip Phone

    Enter the desired values in the Interdigit Long Timer field and the Interdigit Short Timer field. Refer to Step 5 the definitions at the beginning of this section. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 68 Chapter 6 Configure Dial Plan Reset the Control Timers Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 69: Configure Regional Parameters And Supplementary Services

    Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences. Ranges from 0 to 64 seconds. Defaults to 3. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 70: Localize Your Conference Phone

    Defaults to blank; the maximum number of characters is 512. For example: <Language_Selection ua="na"> Spanish </Language_Selection> Locale Choose the locale that should be set in the HTTP Accept-Language header. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 71: Manage The Time And Date

    Configure Regional Parameters and Supplementary Services Localize Your Conference Phone Manage the Time and Date The Cisco conference phone obtains the time settings in one of three ways: NTP Server—When the phone boots up, it tries to contact the first Network Time Protocol (NTP) •...
  • Page 72: Daylight Saving Time Examples

    This section describes how to localize the display language on the conference phone. You can define up to twelve languages, in addition to English, to be available and host the dictionaries for each of the languages on the HTTP or TFTP provisioning server. Language support follows Cisco dictionary principles.
  • Page 73: Create A Dictionary Server Script

    The following languages are supported on the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control: da-DK : Danish_Denmark – nl-NL : Dutch_Netherlands – fr-FR : French_France –...
  • Page 74: Localization Configuration Example

    Localization Configuration Example Language Selection: French (Entry dx must match one of the languages supported by the dictionary server.) Locale: fr-FR (Entry lx must be within the Locale option list.) Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 75: Cisco Unified Ip Conference Phone 8831 For Third-Party Call Control Field Reference

    Displays the primary DNS server assigned to the phone. Current Gateway Displays the default router assigned to the phone. Secondary DNS Displays the secondary DNS server assigned to the phone. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 76: Reboot History

    Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Info Reboot History The conference phone stores the reasons for the last five reboots or refreshes. When the phone is reset to factory defaults, this information is deleted.
  • Page 77: Product Information

    Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Info Product Information Parameter Description Product Name Model number of the conference phone. Software Version Version number of the conference phone software. MAC Address Hardware address of the conference phone.
  • Page 78: Call 1 Status/Call 2 Status

    Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Info Parameter Description Next Registration In Number of seconds before the next registration renewal. Mapped SIP Port Port number of the SIP port mapped by NAT.
  • Page 79 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Info Parameter Description MOS-LQK Score that is an objective estimate of the mean opinion score (MOS) for listening quality (LQK) that rates from 5 (excellent) to 1 (bad). This score is based on audible concealment events due to frame loss in the preceding 8-second interval of the voice stream.
  • Page 80: Download Status

    Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Info Download Status Downloaded Ring Tone Parameter Description Ring Tone Download Status Indicates whether the phone is downloading a ring tone (and from where) or if it is idle.
  • Page 81: Custom Ca Status

    Not Installed—Displays if no custom CA certificate is • installed. Custom CA certificates are configured in the Provisioning tab. For more information about custom CA certificates, see the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Provisioning Guide. Debug Info Console Logs Displays the syslog output of the phone in the reverse order, where messages is the latest one.
  • Page 82: Browser Info

    Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Browser Info Parameter Description Loading Time The amount of elapsed time when the page is loaded on the browser. Safari and IE version before 9 does not support this Note parameter.
  • Page 83: Internet Connection Type

    Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Parameter Description User Password Password for the user. Defaults to blank. Phone-UI-User-Mode Allows you to restrict the menus and options that phone users see when they use the phone interface. Choose yes to enable this parameter and restrict access.
  • Page 84: Vlan Settings

    Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Parameter Description Syslog Server Specify the syslog server name and port. This feature specifies the server for logging IP phone system information and critical events. If both Debug Server and Syslog Server are specified, Syslog messages are also logged to the Debug Server.
  • Page 85 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Parameter Description Network Startup Delay Setting this value causes a delay for the switch to get to the forwarding state before the phone will send out the first LLDP-MED packet.
  • Page 86 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Parameter Description SIP Reg User Agent Name User-Agent name to be used in a REGISTER request. If this is not specified, the <SIP User Agent Name> is also used for the REGISTER request.
  • Page 87 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Parameter Description ReINVITE Expires ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 2000000.
  • Page 88 Interval between NAT-mapping keep alive messages. Defaults to 15. Provisioning For information about the Provisioning page, see the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Provisioning Guide. Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...
  • Page 89 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Regional Control Timer Values (sec) Parameter Description Interdigit Long Timer Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed.
  • Page 90 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Parameter Description Daylight Saving Time Rule Enter the rule for calculating daylight saving time; it should include the start, end, and save values. This rule is comprised of three fields.
  • Page 91 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Localization Parameter Description Dictionary Server Script Defines the location of the dictionary server, the languages available, and the associated dictionary. See the “Create a Dictionary Server Script” section on page 7-5.
  • Page 92 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice General Parameter Description Station Display Name Name to identify the conference phone; appears on the phone screen. You can use spaces in this field and the name does not have to be unique.
  • Page 93 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Miscellaneous Line Key Settings Parameter Description Call Appearances Per Line This parameter allows you to choose the number of calls per line button. You can choose a value from 2 (the default) to 10.
  • Page 94 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Parameter Description Directory User ID BroadSoft User ID of the phone user; for example, johndoe@xdp.broadsoft.com. Directory Password Alphanumeric password associated with the User ID. LDAP Corporate Directory Search...
  • Page 95 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Parameter Description LDAP First Name Filter This defines the search for the common name [cn]. For example, cn:(cn=*$VALUE*). This search allows the provided text to appear anywhere in a name, beginning, middle, or end.
  • Page 96 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice XML Service Parameter Description XML Directory Service Name: Name of the XML Directory. Displays on the user’s phone as a directory choice XML Directory Service URL URL where the XML Directory is located.
  • Page 97 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Voice Supplementary Services Parameter Description Time Format: Choose the time format for the phone (12 or 24 hour). Date Format Choose the date format for the phone (month/day or day/month).
  • Page 98 Appendix A Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Field Reference Call History NAT Settings Parameter Description NAT Keep Alive Enable To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. Defaults to No.
  • Page 99 A P P E N D I X Related Documentation Cisco provides a wide range of resources to help you and your customer obtain the full benefits of the Cisco Unified IP Conference Phone 8831 for Third-Party Call Control. Use the following sections to obtain related information: Cisco IP Phone 8800 Series Documentation, page B-1 •...
  • Page 100 Appendix B Related Documentation Documentation, Service Requests, and Additional Information Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide...