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of Polycom.
®
, the Polycom logo and the names and marks associated with Polycom products are
Disclaimer While Polycom uses reasonable efforts to include accurate and up-to-date information in this document,
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business information), even if Polycom has been advised of the possibility of such damages.
End User License Agreement By installing, copying, or otherwise using this product, you acknowledge that you
have read, understand and agree to be bound by the terms and conditions of the End User License Agreement for this
product. The EULA for this product is available on the Polycom Support page for the product.
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Open Source Software Used in this Product This product may contain open source software. You may receive
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Polycom Support Visit the Polycom Support Center for End User License Agreements, software downloads,
product documents, product licenses, troubleshooting tips, service requests, and more.
The Polycom® Unified Communications (UC) Software Administrator Guide provides general guidance on
installing, provisioning, and managing Polycom phones. This guide helps you understand the Polycom
Voice over Internet Protocol (VoIP) network, and helps you:
● Set up a large-scale provisioning environment
● Set up a VoIP network and provisioning server
● Configure phone features and user settings
● Troubleshoot, update, and maintain phones and UC Software
This Polycom UC Software Administrator Guide applies to the following Polycom devices except where
noted:
● Polycom VVX business media phones, including the VVX 101, 201, 300 series (300/301/310/311),
400 series (400/401/410/411), 500 series (500/501) 600 series (600/601), and 1500
● Polycom VVX Expansion Modules
● Polycom VVX D60 Wireless Handset and Base Station
● Polycom
● UC Software versions 5.3.0 and later do not support use of the VVX 1500 business media phone with
Microsoft Skype for Business Server
SoundStructure® VoIP Interface
Audience and Purpose of This Guide
The primary audience for this guide is the person administering the session initiation protocol (SIP) server,
provisioning servers, VoIP network, and Polycom UC Software that enable you to configure and manage
phone features. This guide is not intended for end users. This guide provides information primarily for
mid-level administrators with experience in networking who understand the basics of open SIP networks
and VoIP endpoint environments. This guide indicates where information might be useful for novice
administrators, and provides tips for advanced administrators where applicable.
Before reading this guide, you should be familiar with the following:
● Computer networking and driver administration for your operating system
● SIP networks
● VoIP environments and technologies
● An XML editor
In addition, this administrator guide provides guidance on the following Polycom-specific skills:
● Polycom provisioning methods
● Polycom UC Software and XML configuration files
Polycom, Inc. 26
Before You Begin
● Configuration parameters and values for end-user device features
● Troubleshooting your Polycom devices
● Maintaining and updating devices and software
Web Info: Latest Polycom UC Software
To find out what’s new for this release of UC Software, including enhanced features, and known and
resolved issues, see the release notes at Latest Polycom UC Software Release.
Phone Deployments
Because phone deployments vary, and administrators typically set up and maintain large-scale device
deployments, Polycom cannot recommend a specific deployment scenario. For large-scale deployments,
Polycom recommends setting up a provisioning server on the local area network (LAN) or on the Internet.
For this reason, this administrator guide focuses on large-scale UC Software VoIP environments set up on
a central SIP and provisioning server. Administrators typically use the administrator guide in three
large-scale device deployment scenarios:
● Enterprisedeployment An administrator sets up and maintains a deployment for a single
organization and all users are in one physical location.
● Multisite enterprise An administrator sets up and maintains a deployment for an organization and
users are spread out over several locations varying in size.
● Service Provider Deployment Service providers provide devices and service to a number of
organizations and users spread out over several locations each varying in size.
What You Need
You require the following to operate Polycom phones as SIP endpoints in large-scale deployments:
● A working IP network
● Routers configured for VoIP
● VoIP gateways configured for SIP
● The latest (or a compatible version) Polycom UC Software image
● An active, configured call server to receive and send SIP messages. For information on IP PBX and
softswitch vendors, see Polycom Desktop Phone Compatibility. If you are using the Polycom
RealPresence Trio Solution, see Polycom RealPresence Trio and SoundStation IP Platform
Compatibility. At minimum, your call server requires:
A call server address that registers voice endpoints with the SIP server
SIP authentication user name and password the phone uses to respond to any SIP authentication
challenges from the SIP server.
● An XML editor—such as XML Notepad—to create and edit configuration files
Get Help
For more information about installing, configuring, and administering Polycom products, refer to Documents
and Downloads at Polycom Support.
To access the latest Polycom UC Software Release Notes, refer to Polycom Voice Support.
Polycom, Inc. 27
Before You Begin
To access the user guide for Polycom voice products, refer to the product support page for your phone at
Polycom Voice Support.
Some Polycom products contain open source software. For details, refer to Polycom Voice Support.
To find help or technical support for your phones, you can search for Polycom documentation at the Polycom
You can find Request for Comments (RFC) documents by entering the RFC number at
http://www.ietf.org/rfc.html.
For other references, look for the Web Info icon throughout this administrator guide.
The Polycom Community
The Polycom Community gives you access to the latest developer and support information and enables you
to participate in discussion forums to share ideas and solve problems with your colleagues. To register with
the Polycom Community, create a Polycom online account. When logged in, you can access Polycom
support personnel and participate in developer and support forums to find the latest information on
hardware, software, and partner solutions topics.
For support or service, please contact your Polycom reseller or visit support.polycom.com for software downloads,
product documents, product licenses, troubleshooting tips, service requests, and more.
We are constantly working to improve the quality of our documentation, and we would appreciate your
feedback. Please send email to VoiceDocumentationFeedback@polycom.com.
Polycom recommends that you record the phone model numbers, software versions (for both the Updater
and UC Software), and partner platform for future reference.
Phone models:
Updater version:
UC Software version:
Partner Platform:
Polycom, Inc. 28
Polycom UC Software Provisioning
Overview
This section provides a high-level overview of Polycom Unified Communications (UC) Software and the
major tasks required to provision Polycom phones with UC Software. Because provisioning environments
vary, Polycom cannot recommend a specific environment. However, Administrators typically set up and
maintain large-scale device deployments and this administrator guide focuses on large-scale UC Software
VoIP environments.
What Is Polycom UC Software?
The Polycom phone software comprises four components:
● Updater The software that loads first when the phone is powered on. For information about the
Updater, see the section About the Updater.
● Polycom UC Software The software that implements the phone functions and features. For
information about Polycom UC Software, see the section About Polycom UC Software.
To begin provisioning devices with Polycom UC Software, refer to Provision with Polycom UC
Software.
● Configuration files The files included with the UC Software download that contain the phone’s
settings. Configuration files are for use with the centralized provisioning method. For information
about the use of configuration files with centralized provisioning, see the section Use Centralized
Provisioning.
● Resource files Optional configuration files that contain settings for advanced features. For
information about the use of configuration files with centralized provisioning, see the section Use
Centralized Provisioning.
About the Updater
The Updater is a small application that resides in the flash memory on the phone. Polycom phones come
installed with the Updater.
Note: The Updater is also known as BootROM
The Updater was referred to as the BootROM in previous versions of the UC Software, specifically UC
Software 3.3.x and SIP 3.2.x and earlier.
When you start/boot/reboot the phone, the Updater automatically performs the following tasks:
1 The setup menu displays so you can set various network and provisioning options.
The Updater requests IP settings and accesses the provisioning server (also called the boot server)
to look for changes to the Updater software. If updates are found, they are downloaded and saved to
flash memory, which overwrites itself after verifying the integrity of the download.
Polycom, Inc. 29
Polycom UC Software Provisioning Overview
2 If new updates are downloaded, the Updater formats the file system, removes any application
software and configuration files that were present.
3 The Updater downloads the master configuration file.
The Updater and the application use this file to acquire a list of other files that the phone needs.
4 The Updater examines the master configuration file for the name of the application file, and then
looks for this file on the provisioning server.
If the copy on the provisioning server is different from the one stored in device settings, or there is no
file stored in flash memory, the application file is downloaded.
5 The Updater extracts the Polycom UC Software from flash memory.
6 The Updater installs the application into RAM, and then uploads an event log file from the boot
cycle.
7 The Updater completes the cycle, and the Polycom UC Software begins running the phone’s
operations.
About Polycom UC Software
Polycom UC Software manages the protocol stack, the digital signal processor (DSP), the user interface,
the network interaction, and implements the following functions and features on the phones:
● VoIP signaling for a wide range of voice and video telephony functions using SIP signaling for call
setup and control.
● SIP and H.323 signaling for video telephony.
● Industry standard security techniques for ensuring that all provisioning, signaling, and media
transactions are robustly authenticated and encrypted.
● Advanced audio signal processing for handset, headset, and speakerphone communications using
a wide range of audio codecs.
● Flexible provisioning methods to support single phone, small business, and large multi-site enterprise
deployments.
● The software is a binary file image and contains a digital signature that prevents tampering or the
loading of rogue software images. There is a new image file in each release of software. Both the
Updater and Polycom UC Software run on all Polycom device models.
UC Software Deployment Scenarios
Administrators typically use the administrator guide in three large-scale device deployment scenarios:
● Enterprisedeployment An administrator sets up and maintains a deployment for a single
organization and all users are in one physical location.
● Multisite enterprise An administrator sets up and maintains a deployment for an organization and
users are spread out over several locations varying in size.
● Service Provider Deployment Service providers provide devices and service to a number of
organizations and users spread out over several locations each varying in size.
Overview of Major Deployment Tasks
A typical large-scale deployment requires administrators to complete each of the following major tasks.
Polycom, Inc. 30
Polycom UC Software Provisioning Overview
To provision phones with UC Software:
1 Create user accounts on the SIP call server.
2 (Optional) Set up a provisioning server. In some cases a provisioning server is built into the SIP call
server and if not, administrators must set up their own provisioning server.
Polycom strongly recommends setting up a provisioning server for large-scale VoIP device
deployments. A provisioning server maximizes the flexibility you have when installing, configuring,
upgrading, and maintaining the phones, and enables you to store configuration, log, directory, and
override files on the server.
3 (Optional) Configure security options on your network.
802.1X
Virtual local area networks (VLANs)
File transfers using HTTPS
Configuration files
SIP signaling over Transport Layer Security (TLS)
4 Set up Dynamic Host Configuration Protocol (DHCP).
5 Set up Domain Name System (DNS). Polycom supports the following DNS records types:
DNS A record
Service (SRV) record for redundancy
Name Authority Pointer (NAPTR)
6 Connect the phones to the network.
The following figure illustrates one example of a device deployment.
Polycom, Inc. 31
Polycom UC Software Provisioning Overview
Polycom wired phones in a network
Polycom, Inc. 32
Provision with Polycom UC Software
After you create user accounts on the SIP call server you are using, provision devices with Polycom UC
Software. This section explains methods you can use to provision devices and configure features on the
phones. Available phone features are listed in the section Configure Devices and Call Controls. You can use
one or multiple methods at the same time but note that features and settings vary by configuration method
and by device.
It is important to be aware that there is a configuration priority among the methods when you use multiple
methods at the same time—settings you make using a higher priority configuration method override settings
made using a lower priority configuration method. When using multiple configuration methods, a setting you
make using a lower-priority method does not apply to or override a duplicate setting made using a
higher-priority method.
● Local device interfaceYou can use the menu system or device interface to provision and
configure one device at a time. Note that menu systems and interface settings vary among devices.
Settings you make from the device menu or interface override settings you make using the Web
Configuration Utility and settings you make on the central provisioning server.
● Web Configuration Utility This method provisions and configures features for one phone at a time
and is recommended for device deployments of fewer than 20 devices. This method enables you to
provision and configure phones using a web browser and enables you to manage phones remotely.
However, note that the Web Configuration Utility contains a limited number of settings. Settings you
make using the Web Configuration Utility override settings you make on the central provisioning
server.
● Centralized provisioning Use this method for large-scale device deployments. This method
requires you to set up your own provisioning server if your SIP call server does not provide one.
Settings you make from a central provisioning server override default device and software settings.
● You can use RPRM version 9.0 or later as a central provisioning server for RealPresence Trio 8800
systems registered with Cisco
RealPresence Resource Manager and RealPresence Trio on Documents & Downloads at Polycom
Support.
®
Unified Communication Manager. For information, see Polycom
Web Info: Registering a single Polycom phone
If you want to register a single Polycom phone, see Quick Tip 44011: Registering Standalone Polycom
SoundPoint IP, SoundStation IP, and VVX 1500 Phones on Polycom Engineering Advisories and
Technical Notifications.
Polycom, Inc. 33
Provision with Polycom UC Software
The following figure illustrates the configuration method override priority among the configuration methods.
Configuration method override priority
Provisioning Points to Consider
● If you are provisioning multiple phones, Polycom recommends that you set up a provisioning server
to install and maintain your Polycom phones, as shown in the section Use Centralized Provisioning.
● A provisioning server maximizes the flexibility you have when installing, configuring, upgrading, and
maintaining the phones, and enables you to store configuration, log, directory, and override files on
the server. You can set up a provisioning server on the local area network (LAN) or anywhere on the
Internet. If you allow the phone write access to your provisioning server, the phone can use the server
to upload all of the file types and store administrator and user settings.
● Polycom phones boot up without the use of configuration files. You can specify a SIP server address
and a registration address (the equivalent of a phone number) in a configuration file before or after
the phone boots up or, after the phone boots up, from the phone’s interface or the Web Configuration
Utility.
● If a phone cannot locate a provisioning server upon boot up, and has not been configured with
settings from any other source, it operates with internally stored default values. If the phone that
cannot locate a provisioning server has previously been configured with settings it operates with
those previous settings.
● Each phone may open multiple connections to the server.
● Settings available only to administrators require a password and are not available to users.
Non-administrative users cannot duplicate or override administrator-level settings.
● To view phone provisioning information, use the multikey shortcut by simultaneously pressing 1-4-7
to display:
Phone IP address
Phone MAC address
VLAN ID
Boot server type (FTP, TFTP, HTTP, HTTPS)
Boot Server Address
Polycom, Inc. 34
Provision with Polycom UC Software
Note: Use RFC-compliant servers
Polycom recommends that you use RFC-compliant servers.
Change Settings from the Device Interface
You can change provisioning settings and phone features locally from the menu system on the phone’s user
interface on a per-phone basis. Settings you make from the device menu or interface override settings you
make using the Web Configuration Utility and settings you make on the central provisioning server. As with
the Web Configuration Utility, the phone device interface makes settings available to users and
administrators; settings available to administrators only can be accessed on the Advanced menu and
require an administrator password. For information on setting passwords, see Set Local User and
Administrator Passwords.
Provision with the Web Configuration Utility
You can provision devices with UC software and control features and settings using the Web Configuration
Utility, a web-based interface that is especially useful for remote configuration. Because features and
settings can vary by device model and UC Software release, options available in the Web Configuration
Utility can vary. The Web Configuration Utility makes settings available to users and administrators; settings
available to administrators only can be accessed on the Advanced menu and require an administrator
password. For information on setting passwords, see Set Local User and Administrator Passwords.
The Web Configuration Utility enables you to perform configuration changes on a per-phone basis. Note
that the Web Configuration Utility contains a limited number of settings you can configure. You can use the
Web Configuration Utility as the sole configuration method or in conjunction with centralized provisioning
and the device interface. If you are provisioning more than ten or twenty phones, Polycom recommends
using centralized provisioning as your primary configuration method.
Configuration changes made to individual phones using the Web Configuration Utility override configuration
settings made with central provisioning. Configuration changes made using a phone’s user interface
override settings made using the Web Configuration Utility. If you want to remove settings applied from the
Web Configuration Utility, click the Reset to Default button on any page in the Web Configuration Utility.
Web Info: Using the Web Configuration Utility
For more detailed help using the Web Configuration Utility, see the Polycom Web Configuration Utility
User Guide on Polycom UC Software Support Center.
Note: Updating UC Software on a single phone
You can use the Software Upgrade tool in the Web Configuration Utility to update the UC Software
version running on a single phone. For information, see Feature Profile 67993: Using the Software Upgrade T ool in the Web Configuration Utility on Polycom Profiled UC Software Features.
Use Centralized Provisioning
This section explains the Polycom UC Software configuration files, and shows you how to set up a
provisioning server.
Polycom, Inc. 35
Provision with Polycom UC Software
The Polycom UC Software that you download contains template configuration files, valid XML files that you
can modify using an XML editor. Use of the configuration files to provision the phones with UC Software and
to modify features and settings is called the centralized provision method. The configuration files enable you
to maintain a set of configuration files for all your devices on a central provisioning server and configure all
of your phones to read the same set of files.
The template configuration files are flexible: you can rearrange the parameters within the template, move
parameters to new files, or create your own configuration files from parameters you want. This flexibility is
especially useful when you want to apply a set of features or settings to separate groups of phones. You
can create and name as many configuration files as you want and your configuration files can contain any
combination of parameters.
Note: Configuration method priority
Remember that settings made from the phone user interface and Web Configuration Utility override
settings you make in configuration files using centralized provisioning.
Centralized provisioning requires that the phone be able to read files and directories you list in the master
configuration file. In addition, the phone attempts to upload log files (log files provide a history of phone
events), a configuration override file, and a provisioning directory file to the provisioning server. Though not
required, Polycom recommends configuring a separate directory for each of these files to help organize: a
log file directory, an override directory, a contact directory, and a license directory.
Each directory can have different access permissions, for example, you can allow log, contacts, and
overrides to have full read and write access, and license to have read-only access. However, where the
security environments permits, Polycom recommends that you allow these file uploads to the provisioning
server which requires you to give delete, write, and read permissions for the phone’s server account. All
other files that the phone needs to read, such as the application executable and the standard configuration
files, should be made read-only using file server file permissions. Ensure that the file permissions you create
provide the minimum required access and that the account has no other rights on the server. Without
permissions, the phone cannot upload files.
Note: Allow file uploads to your provisioning server
Allowing file uploads can help Polycom provide customer support when diagnosing issues with the
phone
In addition to the template configuration files, the UC Software download includes the following configuration
files:
● Language dictionaries
● Ringtones
● Contact directories
If you need to change resource file settings, for example, if you are configuring a phone for a different user,
you need to apply factory default settings to that phone as shown in the section Use an MKC to Reset the
Phone to Defaults
Note that as of Polycom UC Software 4.0.0, you can create user-specific configuration files that enable
phone users to use their features and settings from any phone in an organization. For instructions, refer to
the section Set User Profiles.
The following figure shows one example of a phone network layout when you use the centralized
provisioning method.
Polycom, Inc. 36
Provision with Polycom UC Software
Network layout using centralized prov isioning
Polycom, Inc. 37
Provision with Polycom UC Software
Set Up the Provisioning Server
This section provides instructions for setting up a centralized provisioning server for your Polycom phones.
Polycom phones support the FTP, TFTP, HTTP, and HTTPS protocols, and use FTP by default. The
example shown in this section uses FTP and a personal computer (PC) as the provisioning server.
Prerequisites
To begin, install and set up tools on your PC and gather some information:
● If using Power over Ethernet (PoE) with the phone, you need a PoE switch and network cable.
● Install an XML editor, such as XML Notepad 2007, on your computer.
● Install an FTP server application on your computer. FileZillaand wftpd are free FTP applications for
windows and vsftpd is typically available with all standard Linux distributions.
● Take note of the following:
SIP Server addressThis is the hostname or IP address of the call server that handles VoIP
services on your network.
SIP account information This may include SIP credentials such as a user name and password,
and the phone’s registration address. Although a user name and password are not required to get
the phone working, Polycom strongly recommends using them for security reasons.
MAC address Each phone has a unique 12-digit serial number just above the phone’s bar code
on a label on the back of the phone. Collect the MAC address for each phone in your deployment.
Your computer’s IP address To use your computer as the provisioning boot server, you need
your computer’s IP address. Jot this number down as you need it at the end of the provisioning
process.
To set up the provisioning server:
1 Provide power to the phone using a PoE switch, if available, or, if no PoE switch is available, using
an external power adapter and a network cable to connect the phone to your network.
2 Install and set up an FTP application. FileZillaand wftpd are free FTP applications for windows and
vsftpd is typically available with all standard Linux distributions.
You must create a root FTP directory on the provisioning computer with full read and write access to
all directories and files. You will be placing configuration files in this root directory.
In your FTP server application, create a user account for the phone to use and take note of the user
name and password as you will need these later in the provisioning process. Launch the FTP
application and keep it running at all times so that the phones can communicate with the UC
Software.
3 Download the UC software version(s) to your root directory from the Polycom UC Software Support
Center. To match a phone model with a correct Polycom UC Software release, refer to the Polycom
UC Software Release Matrix for VVX Phones and SoundStructure.
You can choose the combined UC Software package or the split UC Software package, both in ZIP
file format.
The combined version contains all files for all phone models.
The split software package is smaller, downloads more quickly, and contains sip.ld files for each
phone model, enabling you to choose provisioning software for your phone model(s) and maintain
software versions for each model in the same root directory.
4 To apply security settings to your configuration files, refer to the section Encrypt Configuration Files.
Polycom, Inc. 38
Provision with Polycom UC Software
Configure Multiple Servers
You can configure multiple (redundant) provisioning servers—one logical server with multiple addresses—
by mapping the provisioning server DNS name to multiple IP addresses. The default number of provisioning
servers is one and the maximum number is eight. For more information on the protocol used, see Supported
Provisioning Protocols.
If you set up multiple provisioning servers, you must be able to reach all of the provisioning servers with the
same protocol and the contents on each provisioning server must be identical. You can use the table to
configure the number of times each server is tried for a file transfer and also how long to wait between each
attempt. You can configure the maximum number of servers to be tried. For more information, contact your
certified Polycom reseller.
Deploy Devices from the Provisioning Server
After setting up your provisioning server(s), you can deploy your devices. This section shows you how to
deploy your Polycom devices from the provisioning server.
To deploy phones with a provisioning server:
1 Using the list of MAC addresses of each phone you are deploying, create a per-phone
phone<MACaddress>.cfg file.
Do not use the following file names as your per-phone file name: <MACaddress>-phone.cfg,
<MACaddress>-web.cfg, <MACaddress>-app.log, <MACaddress>-boot.log, or
<MACaddress>-license.cfg. These file names are used by the phone to store overrides and logging
information.
2 Add the SIP server registration information and user account information to parameters in the
per-phone file, for example reg.1.address, reg.1.auth.userId, reg.1.auth.password, reg.1.label, reg.1.type.
3 Create a per-site site<location>.cfg file.
For example, add the SIP server or feature parameters such as voIpProt.server.1.address
and feature.corporateDirectory.enabled.
Note: Configuring your phone for local conditions
If SNTP settings are not available through DHCP, you need to edit the SNTP GMT offset, and possibly
the SNTP server address for the correct local conditions. Changing the default daylight savings
parameters might be necessary outside of North America. If the local security policy dictates you
might need to disable the local Web (HTTP) server or change its signaling port (see<httpd/>). To
change the default location settings for user interface language and time and date format (see <lcl/>)
4 Create a master configuration file by performing the following steps:
a Enter the name of each per-phone and per-site configuration file created in steps 2 and 3 in the
CONFIG_FILES attribute of the master configuration file (000000000000.cfg). For help using the
master configuration file, see the section Use the Master Configuration File.
For example, add a reference to phone<MACaddress>.cfg and sip<phonemodelname>.cfg.
b (Optional) Edit the LOG_FILE_DIRECTORY attribute of master configuration file to point to the log
file directory.
Polycom, Inc. 39
Provision with Polycom UC Software
c (Optional) Edit the CONTACT_DIRECTORY attribute of master configuration file to point to the
organization’s contact directory.
(Optional) Edit the USER_PROFILES_DIRECTORY attribute of master configuration file if you
intend to enable the user login feature. For more information, see the section Set User Profiles.
d (Optional) Edit the CALL_LISTS_DIRECTORY attribute of master configuration file to point to the
user call lists.
5 Perform the following steps to configure the phone to point to the IP address of the provisioning
server and set up each user:
a On the phone’s Home screen or idle display, select Settings > Advanced > Admin Settings >
Network Configuration > Provisioning Server. When prompted for the administrative
password, enter 456.
The Provisioning Server entry is highlighted.
b Press the Select soft key.
c Scroll down to Server Type and ensure that it is set to FTP.
d Scroll down to Server Address and enter the IP address of your provisioning server.
e Press the Edit soft key to edit the value and the OK soft key to save your changes.
fScroll down to Server User and Server Password and enter the user name and password of the
account you created on your provisioning server, for example, bill1234 and 1234, respectively.
g Press the Back soft key twice.
h Scroll down to Save & Reboot, and then press the Select soft key.
The phone reboots and the UC Software modifies the APPLICATION APP_FILE_PATH attribute
of the master configuration file so that it references the appropriate sip.ld files.
After this step, the UC Software reads the unmodified APPLICATION APP_FILE_PATH attribute.
Then, the phone sends a DHCP Discover packet to the DHCP server. You can locate this in the
Bootstrap Protocol/option ‘Vendor Class Identifier’ section of the packet which includes the
phone’s part number and the BootROM version. For more information, see the section Parse
Vendor ID Information.
6 Ensure that the configuration process completed correctly:
a On the phone, press Settings (Menu if using a VVX 1500) > Status > Platform > Application >
Main to see the UC Software version and Status > Platform > Configuration
configuration files downloaded to the phone.
b Monitor the provisioning server event log and the uploaded event log files (if permitted). All
configuration files used by the provisioning server are logged.
The phone uploads two logs files to the LOG_DIRECTORY directory: <MACaddress>-app.log
and <MACaddress>-boot.log.
You can now instruct your users to begin making calls.
to see the
Settings: View the phone’s provisioning information
To view phone provisioning information, use the multikey shortcut by simultaneously pressing 1-4-7 to
display:
•Phone IP address
•Phone MAC address
•VLAN ID
•Boot server type (FTP, TFTP, HTTP, HTTPS)
Polycom, Inc. 40
Provision with Polycom UC Software
Override Files
When using a central provisioning server as part of your VoIP environment, you have the option to store the
override file to the phone, or you can permit the phone to upload the override file to the provisioning server
by giving the phone write access to the provisioning server. The advantage of allowing the phone write
access to the provisioning server for override files is that user settings for a phone survive restarts, reboots,
and software upgrades you apply to all phones using a provisioning server. You can also use the override
files to save user custom preferences and to apply specific configurations to a device or device group. If you
permit the phone to upload to the provisioning server, the override file is by default named either <MAC Address>-phone.cfg or <MAC Address>-Web.cfg depending on the whether the change was made from
the phone or Web Configuration Utility respectively.
Note: Priority of Configuration Methods
Changes to settings using a configuration method having a higher priority than another
create an override file that is uploaded to your provisioning server directory. The order of
priority is as follows:
Both override files override settings you make from the provisioning server. The phone uploads an override
file each time a configuration change is made from the phone. If you reformat the phone’s file system, the
override file is deleted.
If you need to clear phone settings and features applied by override files, see the section Use an MKC to
Reset the Phone to Defaults.
Use the Master Configuration File
The centralized provisioning method requires you to use a master configuration file, named
00000000000.cfg in the UC Software download.
You can apply the master configuration file to phones in the following ways:
● To all phones If you are applying the same features and settings to all phones, you can use the
default master configuration file to configure all the phones in a deployment. Note that the phones are
programmed to look first for their own <MACaddress>.cfg file and if a phone does not find a
matching file, it looks next for the default file named 000000000000.cfg. If you do create and use a
per-phone master configuration file, make a copy of the default file and rename it.
● To a phone group or to a single phone If you want to apply features or settings to a group of
phones or to a single phone, make a copy of the default master configuration file and rename it. You
can specify a device group by model or part number using the variable substitutions shown in the
section Create Device Groups.
For single phones, rename the file with a naming scheme that uses the phone’s MAC address
<MACaddress>.cfg. The MAC address, also known as the serial number (SN), is a unique a-f
hexadecimal digit assigned to each phone. Note that you can use only lower-case letters, for
example, 0004f200106c.cfg. You can find the MAC address of a phone on a label on the back of the
phone or on the phone’s menu system at Settings (Menu if using a VVX 1500) > Status > Platform > Phone > S/N:. For more information about naming schemes and efficient provisioning with the
master configuration file, refer to the section Use Variable Substitution.
Polycom, Inc. 41
Provision with Polycom UC Software
● Specify a location You can specify the location of a master configuration file you want the phones
to use, for example, http://usr:pwd@server/dir/example1.cfg. The file name must be at
least five characters long and end with .cfg. If the phone cannot find and download a location you
specify, the phone searches for and uses a per-phone master configuration file and then the default
master configuration file.
Note: Pay attention to per-phone file names
Do not use the following names as extensions for per-phone files: <MACaddress>-phone.cfg,
<MACaddress>-Web.cfg, <MACaddress>-app.log, <MACaddress>-boot.log, or
<MACaddress>-license.cfg. These filenames are used by the phone to store override files and
logging information.
The master configuration file contains a number of default fields, as shown in the following figure.
Default fields in the master configuration file
The following describes the XML field attributes in the master configuration file and the APPLICATION
directories.
● APP_FILE_PATH The path for the UC Software application executable. The default value is sip.ld.
Note that the phone automatically searches for the sip.ld and <part number>.sip.ld. This field can
have a maximum length of 255 characters.
If you want the phone to search for a sip.ld file in a location other than the default or use a different
file name, or both, modify the default. For example, you can specify a URL with its own protocol, user
name, and password: http://usr:pwd@server/dir/sip.ld.
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Provision with Polycom UC Software
● DECT_FILE_PATH The path for the application executable for the Polycom
®
VVX® D60 Wireless
Handset. The default value is 3111-17823-001.dect.ld. When the software for a VVX business media
phone with a paired VVX D60 Base Station is updated, the phone also searches for the dect.ld for
any updates to the base station software.
If you want the phone to search for the 3111-17823-001.dect.ld in a location other than the default or
use a different file name, or both, modify the default. For example, you can specify a URL with its own
protocol, user name, and password: http://usr:pwd@server/dir/3111-17823-001.dect.ld.
● CONFIG_FILES Enter the names of your configuration files here as a comma-separated list. Each
file name has a maximum length of 255 characters and the entire list of file names has a maximum
length of 2047 characters, including commas and white space.
If you want to use a configuration file in a different location or use a different file name, or both, you
can specify a URL with its own protocol, user name and password, for example:
ftp://usr:pwd@server/dir/phone2034.cfg.
Note: Order of the configuration files
The order of the configuration files listed in CONFIG_FILES is significant:
•The files are processed in the order listed (left to right).
•If the same parameter is included in more than one file or more than once in the same file, the
phone uses the first (left) parameter.
● MISC_FILES A comma-separated list of files. Use this to list volatile files that you want phones to
download, for example, background images and ringtone .wav files. The phone downloads files you
list here when booted, which can decrease access time.
● LOG_FILE_DIRECTORY An alternative directory for log files. You can also specify a URL. This
field is blank by default.
● CONTACTS_DIRECTORY An alternative directory for user directory files. You can also specify a
URL. This field is blank by default.
● OVERRIDES_DIRECTORY An alternative directory for configuration overrides files. You can also
specify a URL. This field is blank by default.
● LICENSE_DIRECTORY An alternative directory for license files. You can also specify a URL. This
field is blank by default.
● USER_PROFILES_DIRECTORY An alternative directory for the <user>.cfg files.
● CALL_LISTS_DIRECTORY An alternative directory for user call lists. You can also specify a URL.
This field is blank by default.
● COREFILE_DIRECTORY An alternative directory for Polycom device core files to use to debug
problems. This field is blank by default.
The directories labeled APPLICATION_SPIPXXX indicate phone models that are not compatible with the
latest UC Software version. If you are using any of the phone models listed in these directories, open the
directory for the phone model you are deploying, and use the available fields to provision and configure your
phones.
Use Variable Substitution
You can use the master configuration template file, by default named 000000000000.cfg in the UC Software
files you download, to specify features and settings for single phones and phone groups. This section details
two naming schemes you can use to efficiently provision with the master configuration file. The method you
Polycom, Inc. 43
Provision with Polycom UC Software
use depends on your deployment scenario and understanding both methods helps you to deploy and
manage your phones efficiently.
Method One: Define a Per-Phone MACaddress.cfg File
You can create a MACaddress.cfg file for each phone from the master configuration file template.
The advantage of using this method is a high degree of control over each phone. You can apply
configuration files to phones by adding new files to the CONFIG_FILES field of each phone’s
MACaddress.cfg file. If you want to modify or add settings, go to the configuration files for that phone and
make your changes. If all of the phones in your deployment use the same settings, you can create a single
configuration file for each phone.
This method can require some file management work as you need to create and edit at least two unique
files for each phone in your deployment, namely, a MACaddress.cfg file and one or more configuration files
unique to each phone. You can use the template files or you can create your own files from parameters in
the template files. If you do not want to create a new file, add new parameters to any configuration file
already in the CONFIG_FILES field of a phone’s MACaddress.cfg file.
To create a per-phone MAC address configuration files:
1 Create a MACaddress.cfg file for each phone, replacing 000000000000 with the unique MAC
address of each phone you are configuring, for example 0004f2123456.cfg.
You can find the MAC address of your phone on a label on back of the phone.
2 Create a file for each phone containing information unique to each phone, for example, registration
information. You can use the template files in the UC Software download, or you can create your
own configuration file using parameters from the UC Software template files. Give your files a name
that indicates the file contents or purpose. For example, you might use parameters from the
reg-basic.cfg template file to create a registration file named reg-basic_john_doe.cfg.
3 Enter the name of the file you created in step two in the CONFIG_FILES field of the
MACaddress.cfg file you created in step one for each phone.
4 Save the master configuration file.
Method Two: Use a Variable Substitution
This method enables you to configure all phones using a single master configuration file instead of a
MACaddress.cfg file for each phone. This method follows from the phone’s programmed behavior: the
phone looks first for a file containing its own MAC address and if it cannot find that, uses the default
000000000000.cfg master configuration file.
This method is useful if you need to maintain or modify settings common to all of the phones in your
deployment. To apply a common configuration to all phones, you need only create one new configuration
file and add it to the CONFIG_FILES field of the 000000000000.cfg master file. If you want to add a new
phone to your deployment, you need only create one new file.
For more information on creating phone groups and using variable substitutions, see the section Create
Device Groups.
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Provision with Polycom UC Software
To configure using a variable substitution:
1 Create a file for each phone containing information unique to each phone, for example, registration
information. The name of this file must contain the phone's unique MAC address, for example,
reg-basic_0004f2000001.cfg. Each of these phone-specific configuration files must be
named identically, varying only in the MAC address of each phone.
2 Enter the name of any one of your phone-specific files to the CONFIG_FILES field of the master
configuration file.
3 Modify the file name in the CONFIG_FILES field by replacing the phone-specific MAC address with
the variable [PHONE_MAC_ADDRESS] and include the square brackets. You must enter the
variable in the same place you entered the phone’s MAC address in the phone-specific file.
For example, modify reg-basic_0004f2000001.cfg to
reg-basic_[PHONE_MAC_ADDRESS].cfg.
4 Save the master configuration file.
Create Device Groups
You can create custom device groups by:
● Using a variable in the master configuration file
● Appending a device model or part number to a parameter.
Use a Variable in the Master Configuration File
You can use any of the following variable strings to create custom device groups:
● [PHONE_MODEL]
● [PHONE_PART_NUMBER]
● [PHONE_MAC_ADDRESS]
You can find the MAC address of a phone on a label on the back of the phone or on the phone’s menu at
Menu > Status > Platform > Phone > S/N:. To get the model number or part number of a device, see the
table.
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Provision with Polycom UC Software
Append a Device Model or Part Number to a Parameter
You can customize a set of parameters for a specific device model by appending a device model name or
part number descriptor to a parameter. The part number has precedence over the model number, which has
precedence over the original firmware version.
For example, for a VVX 500, CONFIG_FILES_3111-44500-001=”phone1_3111-44500-001.cfg,
sip_3111-44500-001” overrides CONFIG_FILES_ VVX500=”phone1_ VVX500.cfg, sip_ VVX500.cfg”,
which overrides CONFIG_FILES=”phone1.cfg, sip.cfg”.
The following table lists the product name, model name, and part number mapping for Polycom devices.
Product Name, Model Name, and Part Number
Product NameModel NamePart Number
SoundStructure VoIP Interface SSTRVOIP3111-33215-001
The Polycom UC Software download includes a number of template configuration files containing
configuration parameters. Most configuration parameters are located in only one template file; however,
some are included in two or more files. Remember that configuration files you write to the CONFIG_FILES
field of the master configuration file are read from left to right and the phone uses the file it reads first.
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Provision with Polycom UC Software
Caution: Deprecated configuration parameters
Polycom may deprecate configuration parameters that some organizations may still be using—
deprecated parameters will not work. To view a list of deprecated parameters, see the latest Polycom
UC Software Release Notes on Latest Polycom UC Software Release or check the Release Notes for
earlier software versions on Polycom UC Software Support Center.
The following table lists each template file included with the UC Software download.
device.cfgNetwork Configuration parameters. Refer to the
section Modify Ethernet Settings.
features.cfgFeatures including corporate directory, USB
recording, presence, ACD
firewall-nat.cfgFirewall parameters
H323.cfgH.323 video useTypical Hosted Service Provider using
lync.cfgMicrosoft Skype for Business parametersTypical Microsoft Skype for Business
reg-advanced.cfgAdvanced call server, multi-line phonesTypical Hosted Service Provider
reg-basic.cfgBasic registrationSimple SIP device
region.cfgNon-North American geographiesTypical Hosted Service Provider
sip-basic.cfgBasic call serverSimple SIP device
Typical Hosted Service Provider
Typical IP-PBX
Troubleshooting
Administrative settings
Typical Hosted Service Provider
Typical IP-PBX
VVX 500/501, 600/601, and 1500 for
video calls
environment
Typical IP-PBX
Typical Hosted Service Provider
Typical IP-PBX
Typical Hosted Service Provider
sip-interop.cfgAdvanced call server, multi-line phonesTypical Hosted Service Provider
Typical IP-PBX
site.cfgMulti-site operationsTypical Hosted Service Provider
Typical IP-PBX
techsupport.cfgAvailable by special request from Polycom
Customer Support.
video.cfgVVX 500/501, 600/601, and 1500 videoTypical Hosted Service Provider if using
Polycom, Inc. 47
Use for troubleshooting and debugging
only
VVX 500/501, 600/601, and 1500 for
video calls
Provision with Polycom UC Software
Along with the template files, UC Software includes an XML schema file—polycomConfig.xsd—that
provides information like parameters type (Boolean, integer, string, and enumerated type), permitted values,
default values, and all valid enumerated type values. View this template file with an XML editor.
A string parameter, Boolean, and enumerated parameters are shown in the following figures: String
parameter, Boolean parameter, and Enumerated parameter.
String parameter
Boolean parameter
Enumerated parameter
Change Parameter Values
The configuration parameters available in the UC Software use a variety of values, including Boolean,
integer, enumerated types, and arrays (a table of values). Parameters available in the UC Software are
listed in alphabetical order along with a description, the default value, and permissible values in the section
Configuration Parameters.
Note that the values for Boolean configuration parameters are not case sensitive. The values 0, false,
and off are inter-changeable and supported. The values 1, true, and on are interchangeable and
supported. This Administrator Guide documents only 0 and 1.
The following rules apply when you set a parameter with a numeric value outside of its valid range:
● If the configuration file’s value is greater than the allowable range, the maximum value is used.
● If the configuration file’s value is less than the allowable range, the minimum value is used.
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Provision with Polycom UC Software
● If you insert invalid parameter values into the configuration file, the value is ignored and the default
value is used. Invalid parameters values can occur when enumerated type parameters do not match
a pre-defined value, when numeric parameters are set to a non-numeric values, when string
parameters are either too long or short, or when using null strings in numeric fields. Instances of
invalid values are logged in the phone’s log files.
Note Using blank values and special characters in the configuration files
The UC Software interprets Null as empty; that is, attributeName="".
To enter special characters in a configuration file, enter the appropriate sequence using an XML editor:
● & as &
● ” as "
● ’ as '
● < as <
● > as >
● random numbers as &0x12;
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Configure Network Settings
The UC Software supports the deployment of Polycom phones for your device network:
● As a Session Initiation Protocol (SIP)-based endpoint interoperating with a SIP call server or
softswitch. For more information on SIP, see the section Session Initiation Protocol (SIP).
● As an H.323 video endpoint (Polycom VVX 500/501, 600/601, and 1500 business media phones).
Web Info: Using VVX 1500 phones in a strict H.323 environment
For more information on using VVX 1500 phones in a strict H.323 environment, see Deployment
Guide for the Polycom VVX 1500D Business Media Phone at Polycom VVX 1500 D on Polycom
Support.
Polycom devices operate on an Ethernet local area network (LAN). Local area network design varies by
organization and Polycom phones can be configured to accommodate a number of network designs. This
section shows you several automated and manual ways to configure Polycom phones to operate on a LAN.
Connecting your Polycom phone to the LAN initiates the following startup sequence.
● Only step 1 is required and automatic.
● Steps 2, 3, and 4 are optional as these settings can be manually configured on the device. It is
common to complete step 3 using a DHCP server within the LAN.
Startup sequence:
1 The phone establishes network connectivity.
Wired phones establish a 10M/100M/1000M network link with an Ethernet switch device and do not
function until this link is established. If the phone cannot establish a link to the LAN, an error message
‘Networklink is Down’ displays.
2 (Optional) Apply appropriate security and Quality of Service (QoS) settings.
3 Assign the phone to a VLAN and/or 802.1X authentication.
4 Establish DHCP negotiation with the network and IP address, network addressing options, network
gateway address, and time server.
5 Provisioning server discovery.
This is commonly done using DHCP as part of the previous step. As of UC Software 4.0, the phone
contacts the provisioning server after the phone is operational in order to speed up boot time. You
can disable the provisioning server discovery process as a way of reducing load on a provision
server, for example, after a power failure.
Each step is explained in more detail.
Digest Authentication for Microsoft Internet Information Services
To use digest authentication against the Microsoft Internet Information Services server:
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Configure Network Settings
● Use Microsoft Internet Information Server 6.0 or later.
● Digest authentication needs the user name and password to be saved in reversible encryption.
● The user account on the server must have administrative privileges.
● The wildcard must be set as MIME type; otherwise, the phone will not download *.cfg, *.ld and other
required files. This is because the Microsoft Internet Information Server cannot recognize these
extensions and will return a “File not found” error. To configure wildcard for MIME type, see IIS 6.0
does not serve unknown MIME types.
For more information, see Digest Authentication in IIS 6.0 on Microsoft TechNet.
Establish Phone Connection to the Network
The phones are configured to automatically negotiate the Ethernet rate so that no special configuration is
required. Typical network equipment supports one of the three following Ethernet line rates: 10Mbps,
100Mbps, and 1000Mbps. Though you have the option to change the line rates and/or duplex configuration,
Polycom recommends keeping the default settings. If you do change the settings, make the changes before
connecting your device to the network.
The phone supports two features to prevent Denial of Service (DoS):
● Storm Filtering To change this parameter, see the section Modify Ethernet Settings.
● VLAN Filtering To change this parameter, go to the section Modify VLAN Settings. VLAN filtering
for the VVX business media phones is done by the Linux operating system and cannot be disabled.
Support for Storm and VLAN filtering varies by device.
Apply Security and Quality of Service
You have the option of using several layer-2 mechanisms that increase network security and minimize audio
latency. This section describes each of the network security options.
Set Up VLANs and Wired Devices
You can use a virtual local area network (VLAN) to separate and assign higher priority to a voice LAN as a
way of minimizing latency.
There are several methods you can use to configure the phone to work on a particular VLAN. If the phone
receives a VLAN setting from more than one of the following methods, the priority is as follows (from highest
to lowest):
● LLDP Link Layer Discovery Protocol (LLDP) is a vendor-neutral Layer 2 protocol that allows a
network device to advertise its identity and capabilities on the local network. To change these
parameters, go to Modify VLAN Settings.
● CDP Compatible Cisco Discovery Protocol (CDP) is a proprietary Data Link Layer network
protocol. CDP Compatible follows the same set of rules. To change this parameter, go to Modify
VLAN Settings.
● Static The VLAN ID can be manually set from the phone UI or from a configuration file. To change
this parameter, go to Modify VLAN Settings. This sets the device setting parameter only.
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Configure Network Settings
● DHCP Dynamic Host Configuration Protocol (DHCP) is an automatic configuration protocol used
on IP networks. To change this parameter, go to Modify DHCP Settings. To use DHCP for assigning
VLANs, see the section Assign a VLAN ID Using DHCP. Note that use of DHCP for assigning VLANs
is not standardized and is recommended only if the switch equipment does not support LLDP or CDP
Compatible methods.
Set Up 802.1X Authentication
802.1X authentication is a technology that originated for authenticating Wi-Fi links. It has also been adopted
for authenticating computers within fixed LAN deployments. Multiple Device Authentication is available for
Polycom devices as of UC Software 4.0.0.
Note that when VoIP phones with a secondary Ethernet port are used to connect computers on a network,
the 802.1X authentication process becomes more complex since the computer is not directly connected to
the 802.1X switch. To configure 802.1X, see the section Modify 802.1X Settings.
Web Info: Understand 802.1X
®
For more information on 802.1X authentication, see Introduction to IEEE 802.1X and Cisco
Identity-Based Networking Services (IBNS) at Cisco 802.1X.
See also IEEE 802.1X Multi-Domain Authentication on Cisco Catalyst Layer 3 Fixed Configuration
Switches Configuration Example.
There are three ways to configure 802.1X authentication of devices connected to the PC port of the phone:
● You can configure many switches to automatically trust or accept a VoIP phone based on its MAC
address. This is sometimes referred to as MAC Address Bypass (MAB). Note that the maximum
number of individuals MAC addresses supported on the VVX embedded internal switch is 4,000.
Note that the maximum number of individual MAC addresses supported on the VVX embedded
internal switch is 4,000.
● Some switches support a feature that automatically trust a device that requests a VLAN using the
CDP protocol.
● Some deployments support Multiple Device Authentication (MDA). In this situation, both the phone
and the PC separately authenticate themselves.
In this scenario since the phone is closest to the 802.1X switch, the phone needs to notify the switch
when the PC is disconnected. This can be achieved using an 802.1X EAPOL-Logoff message.
Configure Network Settings Using DHCP
After the phone has established network connectivity it needs to acquire several IP network settings. These
settings are typically obtained automatically from a Dynamic Host Configuration Protocol (DHCP) server.
You have the option to configure IP network settings manually from the phone UI, or to pre-provision using
a device.set capability. If you have never set up a provisioning server before, Polycom recommends
reading the information provided in this section.
Web Info: RFC information on DHCP options
For more information on DHCP options, see RFC 2131 and RFC 2132.
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Configure Network Settings
When making the DHCP request, the phone includes information in Option 60 that can assist the DHCP
server in delivering the appropriate settings to the device. For more information, see Using DHCP Vendor Identifying Options with Polycom Phones: Technical Bulletin 54041 at Polycom Engineering Advisories and
Technical Notifications.
Timesaver: Reducing repetitive data
Polycom recommends using DHCP where possible to eliminate repetitive manual data entry.
The following table details the settings supported through the DHCP menu.
DHCP Network Parameters
ParameterDHCP OptionDHCPDHCP INFORM
IP address
Subnet mask1
IP gateway3
Boot server
address
SIP server
address
SNTP server
address
SNTP GMT offset2
SyslogRefer to the section
-•--•
•--•
•--•
Refer to Modify DHCP
Settings or
Provisioning Server
Discovery.
151 Note: You can
change this value by
changing the device
setting. Refer to
<device/>.
Look at option 42, then
option 4.
Modify Syslog Settings
••-•
•--•
•-••
•-••
Configuration File
(application only)
Device
Settings
DNS server IP
address
DNS INFORM
server IP address
Polycom, Inc. 53
6
6
•--•
•--•
Configure Network Settings
DHCP Network Parameters
DNS domain15
VLAN IDRefer to the section
Modify DHCP Settings
Note: Overriding the DHCP value
You can configure parameter values for the SNTP server address and SNTP GMT offset to override
the DHCP value. Refer tcpIpApp.sntp.address.overrideDHCP for more information.
The CDP Compatibility value can be obtained from a connected Ethernet switch if the switch supports
CDP.
•--•
Warning: Link Layer Discovery Protocol (LLDP) overrides Cisco
Discovery Protocol (CDP). CDP overrides Local FLASH which
overrides DHCP VLAN Discovery.
DHCP Option 43
DHCP Option 60 controls how the phone identifies itself to a DHCP server for Polycom-specific options that
must be returned. If the format for Option 60 is set to RFC 3925, then all Option 43 returned values are
ignored. If the format for Option 60 is set to ASCII string, then the Option 43 response should have a
hexadecimal string value encapsulating sub-options that override any options received outside of DHCP
Option 43.
The following table lists supported DHCP Option 43 individual sub-options and combination sub-options.
DHCP Option 43 Configuration Options
OptionResults
Option 1- subnet maskThe phone parses the value from Option 43
Option 2 - Time offsetThe phone parses the value.
Option 3 - RouterThe phone parses the value.
Option 4 - Time serverThe phone parses the value.
Option 6 - Domain Name ServerThe phone parses the value.
Option 7 - Domain Log serverThe phone parses the value.
Option 15 - Domain NameThe phone parses the value.
Option 42 - Network Time Protocol serverThe phone parses the value.
Option 66 - TFTP Server NameThe phone parses the value.
Sub-options configured in Option 43
Options 1, 2, 3, 4, 5, 6, 7, 15, 42, and 66The phone parses the value.
If you do not have control of your DHCP server or do not have the ability to set the DHCP options, enable
the phone to automatically discover the provisioning server address. You can do this by connecting to a
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Configure Network Settings
secondary DHCP server that responds to DHCP INFORM queries with a requested provisioning server
value. For more information, see RFC 3361 and RFC 3925.
Provisioning Server Discovery
After the phone has established network settings, the phone must discover a provisioning server that
administrators typically use to obtain software updates and configuration settings. If you have never set up
a provisioning server before, Polycom recommends reading the information provided in this section.
The phones support several methods to discover a provisioning server:
● Static You can manually configure the server address from the phone's user interface or the Web
Configuration Utility, or you can pre-provision the phone. The server address is manually configured
from the phone’s user interface, the Web Configuration Utility, or pre-provisioned using device.set
in a configuration file.
● DHCP A DHCP option is used to provide the address or URL between the provisioning server and
the phone.
● DHCP INFORM The phone makes an explicit request for a DHCP option (which can be answered
by a server that is not the primary DHCP server). For more information, see RFC 3361 and RFC
3925.
● Quick Setup This feature offers a soft key that takes the user directly to a screen to enter the
provisioning server address and information. This is simpler than navigating the menus to the
relevant places to configure the provisioning parameters. For more information, see Using Quick
Setup with Polycom Phones: Technical Bulletin 45460 at Polycom Engineering Advisories and
Technical Notifications.
To change these parameters, refer to the section Modify Provisioning Server Settings.
Supported Provisioning Protocols
By default, phones are shipped with FTP enabled as the provisioning protocol. Note that there are two types
of FTP method—active and passive—and UC Software is not compatible with active FTP. You can change
the provisioning protocol by updating the Server Type option. Or, you can specify a transfer protocol in the Server Address, for example, http://usr:pwd@server. The server address can be an IP address, domain
string name, or URL, and can be obtained through DHCP. For more information, refer to the section Modify
Provisioning Server Settings.
Configuration file names in the <MACaddress>.cfg file can include a transfer protocol, for example,
https://usr:pwd@server/dir/file.cfg. If you specify a user name and password as part of the server address
or file name, they are used only if the server supports them. If a user name and password are required but
not specified, the device settings are sent to the server.
The Updater performs the provisioning functions of uploading log files, master configuration files, software
updates, and device setting menu changes. To guarantee software integrity, the Updater downloads only
cryptographically signed Updater or UC Software images. Though UC Software supports digest and basic
authentication when using HTTP/HTTPS, the Updater supports only digest authentication when using
HTTP. When using HTTPS, the phone trusts widely recognized certificate authorities and you can add
custom certificates to the phone. Note that log files are not appended when using TTP or HTTPS.
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Configure Network Settings
Settings: Choosing a valid URL
A URL must contain forward slashes instead of back slashes and should not contain spaces. Escape
characters are not supported. If a user name and password are not specified, the Server User and
Server Password from device settings are used.
Web Info: View trusted certificate authorities
For more information, see Certificate Updates for Polycom UC Software and Using Custom
Certificates with Polycom Phones: EA 17877 at Polycom Engineering Advisories and Technical
Notifications.
As of SIP 3.2, TLS authentication is available. For more information, refer to the section Support Mutual TLS
Authentication.
As of UC Software 4.0.0, 802.1X authentication is available. For more information, refer to the section Set
Up 802.1X Authentication.
Digest Authentication for Microsoft Internet Information Services (IIS)
If you want to use digest authentication against the Microsoft Internet Information Services server:
● Use Microsoft Internet Information Server 6.0 or later.
● Digest authentication needs the user name and password to be saved in reversible encryption.
● The user account on the server must have administrative privileges.
● The wildcard must be set as MIME type; otherwise, the phone will not download *.cfg, *.ld and other
required files. This is because the Microsoft Internet Information Server cannot recognize these
extensions and will return a “File not found” error. To configure wildcard for MIME type, see IIS 6.0
does not serve unknown MIME types.
For more information, see Digest Authentication in IIS 6.0 on Microsoft TechNet.
Modify Phone Network Settings
You have the option to modify phone network settings. This section lists network settings available from the
device interface. Use the soft keys, the arrow keys, and the Select and Delete keys to make changes.
If you have never set up a provisioning server before, Polycom recommends reading the information
provided in this section.
You can update the network configuration parameters at one of two stages:
● During the Updater Phase. The setup menu is accessible during the auto-boot countdown of the
Updater phase of operation. While your phone boots up, press the Cancel soft key, and press the
Setup soft key to launch the setup menu. To access the setup menu, you must enter the
administrator’s password.
● After your phone starts and is running UC Software. The network configuration menu is
accessible from the phone’s main menu. Select Menu > Settings > Advanced > Admin Settings > Network Configuration. To access the Advanced menu, you must enter the administrator’s
password.
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Configure Network Settings
Tip: Changing the default administrator password
Polycom recommends that you change the default administrative password. Refer to the section Set
Local User and Administrator Passwords.
Certain settings are read-only depending on the value of other settings. For example, if the DHCP client
parameter is enabled, the Phone IP Address and Subnet Mask parameters are grayed out or not visible
since the DHCP server automatically supplies these parameters and the statically assigned IP address and
subnet mask is not used.
Tip: Resetting network configurations
The phone default network configuration referred to in subsequent sections can be reset to factory
default settings using the phone’s main menu: Menu > Settings > Advanced > Admin Settings > Reset to Defaults > Reset Device Settings. You can also use a multiple key combination, as shown
in the section Use an MKC to Reboot, Reset to Defaults, Update Logs, Set the Base Profile, View
Phone Details.
Modify Main Menu Settings
You can modify the configuration settings shown in the following table from the setup menu while the phone
boots, or from the phone Administrative Settings menu.
Main Menu
NamePossible Values
Provisioning Menu
Refer to the section Modify Provisioning Server
Settings.
Network Interfaces Menu or Ethernet Menu
Refer to the Modify Ethernet Settings.
TLS Security Menu
Refer to the section Modify TLS Security Settings.
SNTP AddressIP address or domain name string
The Simple Network Time Protocol (SNTP) server the phone obtains the current time from.
GMT Offset-13 through +12
The offset of the local time zone from Greenwich Mean Time (GMT) in half hour increments.
DNS ServerIP address
The primary server the phone directs Domain Name System (DNS) queries to.
DNS AltServerIP address
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Configure Network Settings
Main Menu
The secondary server to which the phone directs DNS queries.
DNS Domain
The phone’s DNS domain.
Hostname
The DHCP client hostname.
Syslog Menu
Refer to the section Modify Syslog Settings.
Quick SetupEnabled, Disabled
If enabled, a QSetup soft key displays on the idle screen when you are in Lines View. When you press this soft
key, a menu displays enabling you to configure the parameters required to access the provisioning server.
Note: The Quick Setup option is not available in the Updater.
Reset to Defaults
There are five ways to reset or clear phone features and settings, including settings from web or local override
files.
Base Profile
Use this to enable Skype for Business-compatible phones to register with Skype for Business Server. When set to
Lync, the phone automatically provisions with the minimum parameters required to register with Skype for
Business Server. You cannot modify or customize the Base Profile.
By default, the Base Profile for normal SKUs is set to Generic.The Base Profile for Lync and Skype for Business
SKUs is Lync.
Domain name string
hostname
Generic, Lync
Modify Provisioning Server Settings
You can modify the configuration settings shown in the table Provisioning Server Menu from the
Provisioning Server menu on the phone.
Note: Change the server user and server password parameters
For security reasons, always change the Server User and Server Password fields from their default
values.
Provisioning Server Menu
NamePossible Values
DHCP Menu
Refer to the section Modify DHCP Settings. Note: This menu is disabled when the DHCP client is disabled.
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Provisioning Server Menu
Server Type0=FTP, 1=TFTP, 2=HTTP, 3=HTTPS,
4=FTPS
The protocol that the phone uses to obtain configuration and phone application files from the provisioning server.
Note: Active FTP is not supported for BootROM version 3.0 or later. Passive FTP is supported. Only implicit FTPS
is supported.
Server AddressIP address or URL
Domain name string or a URL. All addresses can be followed by an optional directory. The address can also be
followed by the file name of a .cfg master configuration file, which the phone uses instead of the default
<MACaddress>.cfg file. The provisioning server to use if the DHCP client is disabled, if the DHCP server does not
send a boot server option, or if the Boot Server parameter is set to Static.
The phone can contact multiple IP addresses per DNS name. These redundant provisioning servers must all use
the same protocol. If a URL is used, it can include a user name and password. For information on how to specify a
directory and use the master configuration file, see the section Use the Master Configuration File.
Note: ":", "@", or "/" can be used in the user name or password if they are correctly escaped using the method
specified in RFC 1738.
Server UserString
The user name requested when the phone logs into the server (if required) for the selected Server Type.
Note: If the Server Address is a URL with a user name, this is ignored.
Server PasswordString
The password requested when the phone logs in to the server if required for the selected Server Type.
Note: If the Server Address is a URL with user name and password, this is ignored.
File Transmit Tries1 to 10 Default 3
The maximum number of attempts to transfer a file. (An attempt is defined as trying to download the file from all IP
addresses that map to a particular domain name.)
Retry Wait0 to 300 seconds Default 1
The minimum amount of time that must elapse before retrying a file transfer. The time is measured from the start of
a transfer attempt, which is defined as the set of upload/download transactions made with the IP addresses that
map to a given provisioning server’s DNS. If the set of transactions in an attempt is equal to or greater than the
Retry Wait value, then there is no further delay before the next attempt is started.
Tag SN to UADisabled, Enabled
If enabled, the phone’s serial number (MAC address) is included in the User-Agent header of HTTP/HTTPS
transfers and communications to the browser.
The default value is Disabled.
Upgrade ServerString
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Configure Network Settings
Provisioning Server Menu
The address/URL that is accessed for software updates requested from the phone’s Web Configuration Utility.
ZTP
See Zero-Touch Provisioning Solution on Polycom Support. Also see
ZTP Frequently Asked Questions.
Disabled, Enabled
Modify DHCP Settings
The DHCP menu is accessible only when the DHCP client is enabled. You can update DHCP configuration
settings shown in the table DHCP Menu.
Note: Multiple DHCP INFORM servers
If multiple DHCP INFORM servers respond, the phone should gather the responses from these DHCP
INFORM servers. If configured for Custom+Option66, the phone selects the first response that
contains a valid custom option value. If none of the responses contain a custom option value, the
phone selects the first response that contains a valid option66 value.
•Option 66The phone looks for option number 66 (string type) in the response received from the DHCP server.
The DHCP server should send address information in option 66 that matches one of the formats described for
Server Addressin the section Modify Provisioning Server Settings.
•CustomThe phone looks for the option number specified by the Boot Server Option parameter (below), and
the type specified by the Boot Server Option Type parameter (below) in the response received from the DHCP
server.
•Static The phone uses the boot server configured through the Server Menu. For more information, see the
section Modify Provisioning Server Settings.
•Custom + Option 66 The phone uses the custom option first or use Option 66 if the custom option is not
present.
Note: If the DHCP server sends nothing, the following scenarios are possible:
•If a boot server value is stored in flash memory and the value is not
used.
•Otherwise the phone sends out a DHCP INFORM query.
If a single DHCP INFORM server responds, this is functionally equivalent to the scenario where the primary
DHCP server responds with a valid boot server value.
If no DHCP INFORM server responds, the INFORM query process retries and eventually times out.
•If the server address is not discovered using DHCP INFORM then the phone contacts the ZTP server if the ZTP
feature is enabled. See ZTP Frequently Asked Questions.
0.0.0.0, then the value stored in flash is
Boot Server Option128 through 254 (Cannot be the same as VLAN ID
Option)
When the Boot Server parameter is set to Custom, this parameter specifies the DHCP option number in which the
phone looks for its boot server.
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Configure Network Settings
DHCP Menu
Boot Server Option Type0=IP Address, 1=String
When the Boot Server parameter is set to Custom, this parameter specifies the type of DHCP option in which the
phone looks for its provisioning server. The IP Address provided must specify the format of the provisioning server.
The string provided must match one of the formats described for Server Address in the section Modify Provisioning
Server Settings.
Option 60 Format0=RFC 3925 Binary, 1=ASCII String
RFC 3925 Binary: Vendor-identifying information in the format defined in RFC 3925.
ASCII String: Vendor-identifying information in ASCII.
For more information, see Using DHCP Vendor Identifying Options with Polycom Phones: Technical Bulletin 54041
at Polycom Engineering Advisories and Technical Notifications.Note: DHCP option 125 containing the RFC 3295 formatted data is sent whenever option 60 is sent. DHCP option
43 data is ignored.
Modify Ethernet Settings
The Ethernet Menu is available only if there are multiple network interfaces to the phone.
Note: LAN port mode
You can set the LAN Port Mode on all phones. The PC Port Mode parameters are available only on
phones with a second Ethernet port.
The following phones support the LAN Port Mode and PC Port Mode setting of 1000FD: VVX
310/311, 410/411, 500/501, 600/601, and 1500.
Ethernet Menu
NamePossible Values
DHCPEnabled, Disabled
If enabled, DHCP is used to obtain the parameters discussed in the section Modify DHCP Settings.
IP AddressIP address
The phone’s IP address. Note: This parameter is disabled when DHCP is enabled.
Subnet MaskSubnet mask
The phone’s subnet mask. Note: This parameter is disabled when DHCP is enabled.
IP Gateway
The phone’s default router.
IP address
VLAN
See the section Modify VLAN Settings.
802.1X AuthenticationEnabled, Disabled
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Configure Network Settings
Ethernet Menu
If enabled, the phone uses the 802.1 Authentication parameters to satisfy the negotiation requirements for each
EAP type.
802.1X Menu
See the section Modify 802.1X Settings.
Storm FilteringEnabled, Disabled
If enabled, received Ethernet packets are filtered so that the TCP/IP stack does not process bad data or too much
data. The default value is Enabled.
LAN Port Mode0 = Auto, 1 = 10HD, 2 = 10FD, 3 = 100HD,
4 = 100FD, 5 = 1000FD
The network speed over Ethernet. The default value is Auto. HD means half duplex and FD means full duplex.
Note: Polycom recommends that you do not change this setting.
PC Port Mode0 = Auto, 1 = 10HD, 2 = 10FD, 3 = 100HD,
4 = 100FD, 5 = 1000FD, -1 = Disabled
The network speed over Ethernet. The default value is Auto. HD means half duplex and FD means full duplex.
Note: Polycom recommends that you do not change this setting unless you want to disable the PC port.
Modify VLAN Settings
You can modify the settings listed in the following table.
VLAN Menu
NamePossible Values
VLAN IDNull, 0 through 4094
The phone’s 802.1Q VLAN identifier. The default value is Null. Note: Null = no VLAN tagging
LLDPEnabled, Disabled
If enabled, the phone uses the LLDP protocol to communicate with the network switch for certain network
parameters. Most often this is used to set the VLAN that the phone should use for voice traffic. It also reports power
management to the switch. The default value is Enabled.
For more information on how to set VLAN and LLDP, refer to the section LLDP and Supported TLVs.
CDP CompatibilityEnabled, Disabled
If enabled, the phone uses CDP-compatible signaling to communicate with the network switch for certain network
parameters. Most often this is used to set the VLAN that the phone should use for Voice Traffic, and for the phone
to communicate its PoE power requirements to the switch. The default value is Enabled.
•Fixed: Use predefined DHCP vendor-specific option values of 128, 144, 157 and 191. If one of these is used,
VLAN ID Option is ignored.
•Custom: Use the number specified for VLAN ID Option as the DHCP private option value.
For a detailed description, refer to the section Assign a VLAN ID Using DHCP.
VLAN ID Option128 through 254 (Cannot be the same as Boot Server
Option) (default is 129)
The DHCP private option (when VLAN Discovery is set to Custom).
For a detailed description, refer to the section Assign a VLAN ID Using DHCP.
Modify 802.1X Settings
The 802.1X Menu displays on the VVX 500/501 and 600/601 when 802.1X authentication is enabled.
You can modify configuration parameters shown in the following table.
802.1X Menu
NamePossible Values
EAP Method0 = None, 1=EAP-TLS, 2=EAP-PEAPv0/MSCHAPv2,
A flag to determine whether or not to delete the PAC file from the phone.
Modify Login Credential Settings
You can modify settings shown in the following table.
Login Credentials Menu
NamePossible Values
Domain
The domain name used by a server.
User
The user name used to authenticate to a server.
Password
The password used to authenticate to a server.
UTF-8 encoded string
UTF-8 encoded string
UTF-8 encoded string
Modify TLS Security Settings
This section refers to the TLS Security menu available in the Updater and UC Software. You can modify the
settings shown in the following table.
TLS Security Menu
NamePossible Values
OCSPEnabled, Disabled
The Online Certificate Status Protocol checks the revocation status of X.509 digital certificates downloaded during
negotiation of a TLS connection.
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Configure Network Settings
TLS Security Menu
FIPSEnabled, Disabled
The Federal Information Processing Standard enables the validation and usage of FIPS-140 certified encryption
algorithms.
Install Custom CA CertURL
A CA certificate that is installed on the phone to be used for TLS authentication.
Install Custom Device CertURL
A device certificate installed on the phone to be used for Mutual TLS authentication.
Clear CertificateYes, No
A flag to determine whether or not the device certificate can be removed from the phone.
TLS Profile x
There are currently two TLS Platform profiles. See the section Modify TLS Profile Settings.
Web Services
See the section Modify Applications Settings.
Modify TLS Profile Settings
You can modify settings shown in the following table.
TLS Profile Menu
NamePossible Values
SSL Cipher Suite
The global cipher suite.
Custom SSL Cipher Suite
A custom cipher suite.
CA Cert ListString
The CA certificate sources that are valid for this profile.
Device Cert ListString
The device certificate sources that are valid for this profile.
String
String
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Configure Network Settings
Modify Applications Settings
You can modify settings shown in the following table.
Applications Menu
NamePossible Values
802.1X
The TLS Profile to use for 802.1X authentication.
Provisioning
The TLS Profile to use for provisioning authentication.
ProvisioningEnable or Disable
The TLS Profile to enable or disable common name validation.
Syslog
The TLS Profile to use for syslog authentication.
1 or 2
1 or 2
1 or 2
Modify Syslog Settings
Syslog is a standard for forwarding log messages in an IP network. The term syslog is often used for both
the actual syslog protocol as well as the application or library sending syslog messages.
The syslog protocol is a simple protocol: the syslog sender sends a small textual message (less than 1024
bytes) to the syslog receiver. The receiver is commonly called syslogd, syslog daemon, or syslog server.
Syslog messages can be sent through UDP, TCP, or TLS. The data is sent in cleartext.
Because syslog is supported by a wide variety of devices and receivers, syslog can be used to integrate log
data from many different types of systems into a central repository.
Web Info: Information on Syslog
For more information on the syslog protocol, see RFC 3164.
You can modify settings shown in the following table.
Syslog Menu
NamePossible Values
Server Address
The syslog server IP address. The default value is Null.
Server TypeNone=0, UDP=1, TCP=2, TLS=3
The protocol that the phone uses to write to the syslog server. If set to None (or 0), transmission is turned off, but
the server address is preserved.
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IP address or domain name string
Configure Network Settings
Syslog Menu
Facility0 to 23
A description of what generated the log message. For more information, see section 4.1.1 of RFC 3164.
The default value is 16, which maps to local 0.
Render Level0 to 6
Specifies the lowest class of event that rendered to syslog. It is based on log.render.level and can be a lower
value. See <log/>. Note: Use left and right arrow keys to change values.
Prepend MAC AddresEnabled, Disabled
If enabled, the phone’s MAC address is prepended to the log message sent to the syslog server.
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Configure Devices and Call Controls
This section provides information to successfully configure the UC Software configuration files and
parameters that controls device features and settings.
How to Read the Feature Parameter Tables
Each of the feature descriptions includes a table of parameters that you configure to make the features work.
The following section explains the conventions used in the feature parameter tables.
The feature parameter tables indicate the parameters you can configure for a feature when using the
centralized provisioning method. The following illustration shows you how to read the feature parameter
tables to locate the function of the parameter, the name of the template file the parameter is in, and the name
of the parameter you configure. Each parameter listed in the feature parameter tables is linked to the
reference section Configuration Parameters in this guide, which provides a full description, permissible, and
default values for each parameter.
Feature parameter table format
To locate a parameter in your UC Software download, open the template name indicated. Then use the
parameter name to navigate the folders in the XML tree structure. The parameter name contains the XML
folder path. The two following examples explain this convention in more detail.
Example One: Feature Parameter Ta bles
The following example is taken from the section Call Forward on Shared Lines.
Feature parameter table for multiple call appearances
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Configure Devices and Call Controls
This example indicates that the reg-basic.cfg template file contains the calls.callsPerLineKey
parameter, which sets the number of concurrent calls for a phone’s line keys. Because the default value
varies by device, click on the blue parameter name to go to the parameter description, which includes
default and permissible values. If you want to change the parameter value, locate and open the reg.basic
template, expand the reg folder, and locate the parameter calls.callsPerLineKey. Set the parameter
value to a permissible number, as shown in the following illustration.
Example Multiple Call Appearances
Some file paths in the templates are long and you may have to expand several folders in the XML tree
structure to locate a specific parameter.
Note also that some feature parameters are located in more than one template file. In these cases, the
parameter tables list all related template files.
Note: Each parameter is linked
Each parameter listed in the tables in various sections is linked to its definition in the section
Configuration Parameters. This section defines each parameter and list the permissible values,
including the default value, of each parameter. If you want to find out more about a parameter you see
listed in the tables, click the blue parameter name.
Example Two: Configuring Grouped Parameters
Some of the features have several related parameters that you need to configure to get the feature working.
In these cases, instead of listing every parameter, the table specifies a group of related parameters with an
abbreviated XML path name ending with (.*), which indicates you can configure a group of related
parameters.
Abbreviated XML paths, like full parameter names, are linked to their definitions in the reference sections in
the section Configuration Parameters. Specifically, since the reference sections lists parameters
alphabetically, abbreviated XML path are linked to the first of a group of parameters listed alphabetically in
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Configure Devices and Call Controls
the reference section. The next example shows you that in the site.cfg template, the tcpIpApp.sntp
folder contains several related parameters that configure basic SNTP settings.
Feature parameter table for time and date SNTP settings
This example indicates that there is a group of SNTP parameters you can configure in the site.cfg template
file. The abbreviated parameter name tcpIpApp.sntp.* indicates that you can configure parameters in
the tcpIpApp.sntp folder as well as parameters in tcpIpApp.sntp subfolders.
To locate these parameters in the XML file, use the parameter name. The parameter name contains the XML
folder path, as shown in the following illustration.
Locating parameters in the templates
Note: Using an XML editor
Polycom recommends using an XML editor such as XML Notepad 2007 to open and edit the
configuration template files.
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Configure Devices and Call Controls
Configure Phone Signaling
This section provides information on configuring phone signaling.
Quick Setup of Polycom Phones
A quick setup feature simplifies the process of entering the provisioning (boot) server parameters from the
phone’s user interface. This feature is designed to make it easier for on-site out of the box provisioning of
VVX business media phones.
When you enable this feature, a QSetup soft key displays. When you press the QSetup soft key, a new menu
displays. The menu enables you to access the provisioning server and quickly configure the phone to work.
After configuring the quick setup, you can disable display of the QSetup soft key using a configuration file
setting. The following table indicates the parameter that enables this feature.
Web Info: Configure quick setup
For details on how to configure quick setup, see Technical Bulletin 45460: Using Quick Setup with
To display a QSetup soft key on the phone screen and access the Quick Setup menu, enable the
prov.quickSetup.enabled parameter in the site.cfg template file, as shown next.
Quick Setup Configuration
Configure Real-Time Transport Protocol (RTP) Ports
You can configure the phone to filter incoming RTP packets. You can filter the packets by IP address, or by
port. For greater security, you can also configure RTP settings to reject packets arriving from a
non-negotiated IP address or from an unauthorized source. You can reject packets that the phone receives
from a non-negotiated IP address or a non-negotiated port.
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Configure Devices and Call Controls
You can configure the phone to enforce symmetric port operation for RTP packets. When the source port is
not set to the negotiated remote sink port, arriving packets can be rejected.
You can also fix the phone’s destination transport port to a specified value regardless of the negotiated port.
This can be useful for communicating through firewalls. When you use a fixed transport port, all RTP traffic
is sent to and arrives on that specified port. Incoming packets are sorted by the source IP address and port,
which allows multiple RTP streams to be multiplexed.
You can specify the phone’s RTP port range. Since the phone supports conferencing and multiple RTP
streams, the phone can use several ports concurrently. Consistent with RFC 1889, the next-highest
odd-numbered port is used to send and receive RTP. The following table provides a link to the reference
section.
The phone is compatible with RFC 1889 - RTP: A Transport Protocol for Real-Time Applications - and the
updated RFCs 3550 and 3551. Consistent with RFC 1889, the phone treats all RTP streams as
bi-directional from a control perspective and expects that both RTP endpoints negotiate the respective
destination IP addresses and ports. This allows real-time transport control protocol (RTCP) to operate
correctly even with RTP media flowing in only a single direction, or not at all.
Configure Real-Time Transport Protocol Ports
Parameter Functiontemplate > parameter
Filter RTP packets by IP address.site.cfg > tcpIpApp.port.rtp.filterByIp
Filter RTP packets by port.site.cfg > tcpIpApp.port.rtp.filterByPort
Force-send packets on a specified port.site.cfg > tcpIpApp.port.rtp.forceSend
Set the starting port for RTP packet port range.site.cfg >tcpIpApp.port.rtp.mediaPortRangeStart
Example Real-Time Transport Protocol Configuration
The following illustration shows the default real-time transport protocol settings in the site.cfg template file.
The parameter tcpIpApp.port.rtp.filterByIp is set to 1 so that the phone rejects RTP packets sent
from non-negotiated IP addresses. The parameter tcpIpApp.port.rtp.filterByPort is set to 0 so
that RTP packets sent from non-negotiated ports are not rejected. Enter a value in the
tcpIpApp.port.rtp.forceSend parameter to specify the port that all RTP packets are sent to and
received from. The parameter tcpIpApp.port.rtp.mediaPortrangeStart shows the default starting
port 2222 for RTP packets. The starting port must be entered as an even integer.
Default real-time transport protocol
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Configure Devices and Call Controls
Configure Network Address Translation
The phone can work with certain types of Network Address Translation (NAT). NAT enables a local area
network (LAN) to use one set of IP addresses for internal traffic and another set for external traffic. The
phone’s signaling and RTP traffic use symmetric ports. You can configure the external IP address and ports
used by the NAT on the phone’s behalf on a per-phone basis. The following table lists each of the
parameters you can configure. Note that the source port in transmitted packets is the same as the
associated listening port used to receive packets.
Network Access Translation
Parameter Functiontemplate > parameter
Specify the external NAT IP address.sip-interop.cfg > nat.ip
Specify the external NAT keepalive interval.sip-interop.cfg > nat.keepalive.interval
Specify the external NAT media port start.sip-interop.cfg > nat.mediaPortStart
Specify the external NAT signaling port.sip-interop.cfg > nat.signalPort
Example Network Address Translation Configuration
The following illustration shows the default NAT parameter settings. The parameter nat.ip is the public IP
that you want to advertise in SIP signaling. The default IP is 120.242.6.155.
The parameter nat.mediaPortStart is the RTP used to send media. If non-Null, this attribute is set the
initially allocated RTP port and overrides the value set in
tcpIpApp.port.rtp.mediaPortRangeStart. In the example, the starting port is 12500 and the phone
cycles through start-port + 47 for phones that support audio only or start-port + 95 for phones that support
video.
The parameter nat.signalPort specifies the port that the phone uses for SIP signaling. This parameter
overrides voIpProt.local.Port. In the example below, the phone uses port 5070 for SIP traffic.
Use the nat.keepalive.interval to specify the keepalive interval in seconds. This parameter sets the
interval at which phones sends a keepalive packet to the gateway/NAT device. The keepalive packet keeps
the communication port open so that NAT can continue to function as initially set up. In the example below,
the phone sends the keepalive every 120 seconds.
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Default NAT parameter settings
DNS SIP Server Name Resolution
If a DNS name is given for a proxy/registrar address, the IP addresses associated with that name is
discovered as specified in RFC3263. If a port is given, the only lookup is an A record. If no port is given,
NAPTR and SRV records are tried before falling back on A records if NAPTR and SRV records return no
results. If no port is given, and none is found through DNS, port 5060 is used. If the registration type is TLS,
port 5061 is used.
Caution: No DNS resolution causes failover
Failure to resolve a DNS name is treated as signaling failure that causes a failover.
The following configuration causes the phone to build an SRV request based on the address you provide,
including all subdomains. Use the format:
This SRV request produces a list of servers ordered by weight and priority, enabling you to specify
subdomains for separate servers, or you can create partitions of the same system. Please note that while
making SRV queries and transport is configured as TCP, the phone adds the prefix <_service._proto.>
to the configured address/FQDN but does not remove the subdomain prefix, for example
sip.example.com becomes _sip._tcp.sip.example.com. A single SRV query can be resolved into
many different servers, session border controllers (SBCs), or partitions ordered by weight and priority, for
example, voice.sip.example.com and video.sip.example.com. Alternatively, use DNS NAPTR to
discover what services are available at the root domain.
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Behavior When the Primary Server Connection Fails
For Outgoing Calls (INVITE Fallback)
When the user initiates a call, the phone completes the following steps to connect the call:
1 The phone tries to call the working server.
2 If the working server does not respond correctly to the INVITE, the phone tries and makes a call
using the next server in the list (even if there is no current registration with these servers). This could
be the case if the Internet connection has gone down, but the registration to the working server has
not yet expired.
3 If the second server is also unavailable, the phone tries all possible servers (even those not
currently registered) until it either succeeds in making a call or exhausts the list at which point the
call fails.
At the start of a call, server availability is determined by SIP signaling failure. SIP signaling failure depends
on the SIP protocol being used:
● If TCP is used, then the signaling fails if the connection fails or the Send fails.
● If UDP is used, then the signaling fails if ICMP is detected or if the signal times out. If the signaling
has been attempted through all servers in the list and this is the last server, then the signaling fails
after the complete UDP timeout defined in RFC 3261. If it is not the last server in the list, the maximum
number of retries using the configurable retry timeout is used. For more information, see <server/>
and <reg/>.
Caution: Use long TTLs to avoid DNS timeout delays
If DNS is used to resolve the address for Servers, the DNS server is unavailable, and the TTL for the
DNS records has expired, the phone attempts to contact the DNS server to resolve the address of all
servers in its list before initiating a call. These attempts timeout, but the timeout mechanism can cause
long delays (for example, two minutes) before the phone call proceeds using the working server. To
prevent this issue, long TTLs should be used. Polycom recommends deploying an on-site DNS server
as part of the redundancy solution.
Phone Configuration
The phones at the customer site are configured as follows:
● Server 1 (the primary server) is configured with the address of the service provider call server. The
IP address of the server(s) is provided by the DNS server, for example:
reg.1.server.1.address=voipserver.serviceprovider.com.
● Server 2 (the fallback server) is configured to the address of the router/gateway that provides the
fallback telephony support and is on-site, for example: reg.1.server.2.address=172.23.0.1.
Note: Caution when using multiple servers per registration
It is possible to configure the phone for more than two servers per registration but ensure that the
phone and network load generated by registration refresh of multiple registrations does not become
excessive. This is of particular concern when a phone has multiple registrations with multiple servers
per registration and some of these servers are unavailable.
Phone Operation for Registration
After the phone has booted up, it registers to all configured servers.
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Server 1 is the primary server and supports greater SIP functionality than other servers. For example,
SUBSCRIBE/NOTIFY services used for features such as shared lines, presence, and BLF is established
only with Server 1.
Upon the registration timer expiry of each server registration, the phone attempts to re-register. If this is
unsuccessful, normal SIP re-registration behavior (typically at intervals of 30 to 60 seconds) proceeds and
continues until the registration is successful (for example, when the Internet link is again operational). While
the primary server registration is unavailable, the next highest priority server in the list serves as the working
server. As soon as the primary server registration succeeds, it returns to being the working server.
Note: Failover to servers that are not registered
If reg.x.server.y.register is set to 0, the phone does not register to that server. However, the
INVITE fails over to that server if all higher priority servers are down.
Recommended Practices for Fallback Deployments
In situations where server redundancy for fallback purpose is used, the following measures should be taken
to optimize the solution:
● Deploy an on-site DNS server to avoid long call initiation delays that can result if the DNS server
records expire.
● Do not use OutBoundProxy configurations on the phone if the OutBoundProxy could be unreachable
when the fallback occurs.
● Avoid using too many servers as part of the redundancy configuration as each registration generates
more traffic.
● Educate users as to the features that are not available when in fallback operating mode.
Note: Compatibility with Microsoft Skype for Business
The concurrent/registration failover/fallback feature is not compatible with Microsoft environments.
Configure the Static DNS Cache
Failover redundancy can only be used when the configured IP server hostname resolves (through SRV or
A record) to multiple IP addresses. Unfortunately, the DNS cache cannot always be configured to take
advantage of failover redundancy.
The solution in SIP 3.1 is to enable you to statically configure a set of DNS NAPTR SRV and/or A records
into the phone. See the table Configuring the Static DNS Cache for configurable parameters.
Phones configured with a DNS server behave by default as follows:
● The phone makes an initial attempt to resolve a hostname that is within the static DNS cache. For
example, a query is made to the DNS if the phone registers with its SIP registrar.
● If the initial DNS query returns no results for the hostname or cannot be contacted, then the values
in the static cache are used for their configured time interval.
● After the configured time interval has elapsed, a resolution attempt of the hostname again results in
a query to the DNS.
● If a DNS query for a hostname that is in the static cache returns a result, the values from the DNS
are used and the statically cached values are ignored.
Phones not configured with a DNS server behave by default as follows:
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● When the phone attempts to resolve a hostname within the static DNS cache, it always returns the
results from the static cache.
Support for negative DNS caching as described in RFC 2308 is also provided to allow faster failover when
prior DNS queries have returned no results from the DNS server. For more information, see RFC2308.
Configuring the Static DNS Cache
Parameter Functiontemplate > parameter
Specify the line registration.sip-interop.cfg > reg.x.address
Specify the call server used for this registration.sip-interop.cfg > reg.x.server.y.*
Specify the DNS A address, hostname, and cache time interval.site.cfg > dns.cache.A.x.*
Specify the DNS NAPTR parameters, including: name, order, preference,
regexp, replacement, service, and ttl.
Specify DNS SRV parameters, including: name, port, priority, target, ttl, and
weight.
Specify whether to use DNS primary and secondary address set using the
parameters tcpIpApp.dns.server and tcpIpApp.dns.altServer.
Specify whether to use the DNS domain name set using the parameter
tcpIpApp.dns.domain.
.* indicates grouped parameters. See the section Example Two: Configuring Grouped Parameters for more
information.
site.cfg > dns.cache.NAPTR.x.*
site.cfg > dns.cache.SRV.x.*
site.cfg >
tcpIpApp.dns.address.overrideDH
CP
site.cfg >
tcpIpApp.dns.domain.overrideDH
CP
Example Static DNS Cache Configuration
The following examples show you how to configure the static DNS cache.
Example 1
This example shows how to configure static DNS cache using A records IP addresses in SIP server address
fields.
When the static DNS cache is not used, the site.cfg configuration looks as follows:
When the static DNS cache is used, the site.cfg configuration looks as follows:
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Note: Order of addresses
The addresses listed in this example are read by Polycom UC Software in the order listed.
Example 2
This example shows how to configure static DNS cache where your DNS provides A records for
reg.x.server.x.address but not SRV. In this case, the static DNS cache on the phone provides SRV
records. For more information, see RFC 3263.
When the static DNS cache is not used, the site.cfg configuration looks as follows:
When the static DNS cache is used, the site.cfg configuration looks as follows:
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Note: Port value settings
The reg.1.server.1.port and reg.1.server.2.port values in this example are set to null to
force SRV lookups.
Example 3
This example shows how to configure static DNS cache where your DNS provides NAPTR and SRV records
for reg.x.server.x.address.
When the static DNS cache is not used, the site.cfg configuration looks as follows:
When the static DNS cache is used, the site.cfg configuration looks as follows:
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Settings: Forcing NAPTR lookups
The reg.1.server.1.port, reg.1.server.2.port, reg.1.server.1.transport, and
reg.1.server.2.transport values in this example are set to null to force NAPTR lookups.
Enable Access URL in SIP Messages
When this feature is enabled, the server can attach a URL to incoming and active calls. The phone’s browser
or microbrowser can read this URL and render it as web content that displays on the phone screen. This
feature can be enabled on VVX 500/501 and 1500 phones as shown in the table Enable Access URL in SIP
Messages.
This feature is flexible and can be used in the following ways.
● A Call Center
A URL is attached to an incoming call and displays extended information about a customer before
the agent takes the call.
The phone can display a script of questions for an agent to ask a caller, and a different script can
be provided to different agent groups.
● A Restaurant menu on a hotel phone
A guest dials a number for the restaurant or room service and a voice indicates that the menu is
available for viewing on the phone.
There are three user interface aspects to this feature:
● Web Content Status Indication—When valid web content is available on the phone, an icon
displays beside the call information. In the examples shown next, a lightning bolt icon indicates web
content is available for a call appearance. The phone user can press the Select key to display the
web content.
● Web Content Retrieval—Phone users can choose to retrieve web content in Active Mode
(spontaneously) or in Passive Mode (by request).
Active Mode—There are two ways to configure spontaneous web content retrieval: you can set
the web content retrieval parameter in the configuration file to ‘active’ or, if your call server
supports access URL, you can specify active retrieval in the SIP heading. If parameters in the SIP
signal conflict with the file configuration, parameters in the SIP signaling takes precedence. Note
that incoming active web content interrupts web content currently being viewed.
Passive Mode—There are two ways to configure web content retrieval by request: you can set
the web content retrieval parameter in the configuration file to ‘passive’ or, if your call server
supports access URL, you can specify passive retrieval in the SIP heading. When passive mode
is enabled, an icon displays beside a call appearance indicating that web content is available. For
more information about the web content icon, see Web Content Status Indication earlier in this
section. When an icon shows that web content is available, the phone user can press the Select
key to view the content. If the web content has expired, no icon displays and the Select key
performs no function. Note that incoming active web content interrupts web content currently
being viewed. Passive mode is recommended when the microbrowser is used for other
applications.
● Settings Menu—You can enable new web content to be added to the phone’s menu. Using the
phone's menu, users can set the default display mode for individual URLs to active or passive.
You must use the following standards if you want to set the retrieval display mode of web content in the SIP
headers:
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A new SIP header must be used to report web content associated with SIP phone calls (the
SSAWC header follows the BNF for the standard SIP header Alert-Info):
This header may be placed in SIP requests and responses so long as the messages are part of an
INVITE-initiated dialog and the phone can associate them with an existing phone call.
You may also define two optional parameters:
An expires parameter is defined to indicate the lifespan of the URL itself. Or, if the URL is
permanent, you can set how long the web content displays with the call. An absent or invalid
parameter is interpreted to mean that the content or the URL itself is persistent. A value, if present,
indicates the lifespan of the content in seconds (zero has special significance—see the next
example). When the lifespan expires, the phone removes both the indication of the URL and the
ability of the user to retrieve it.
If the server wishes to invalidate a previous URL, it can send a new header (through UPDATE)
with expires=0. The expires parameter is ignored when determining whether to spontaneously
retrieve the web content unless expires=0.
A mode parameter is defined to indicate whether the web content should be displayed
spontaneously or retrieved on-demand. Two values are allowed: active and passive. An absent
or invalid parameter is interpreted the same as passive, meaning that the web content is
retrievable on-demand but is not be spontaneously displayed. If the value is set to active, the web
content is spontaneously displayed, subject to the rules discussed under Active Mode in Web
Content Retrieval earlier in this section.
In this case, an icon indicates that web content is available for a period of 60 seconds.
Enable Access URL in SIP Messages
Parameter Functiontemplate > parameter
To turn this feature on or off.features.cfg > mb.ssawc.enabled
To retrieve content.features.cfg > mb.ssawc.call.mode
Example Access URL in SIP Messages Configuration
In the following example, in the features.cfg template, the access URL in SIP message feature is enabled
in mb.ssawc.enabled. The parameter mb.ssawc.call.mode is set to passive, which means web
content does not display spontaneously; web content displays when activated by the phone user.
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Access URL in SIP messages example
Display SIP Header Warnings
The warning field from a SIP header may be configured to display a three second pop-up message on the
phone, for example, that a call transfer failed due to an invalid extension number. For more information, refer
to the section Supported SIP Request Headers.
You can display these pop-up messages in any language supported by the phone. The messages display
for three seconds unless overridden by another message or action. To turn the warning display on or off or
specify which warnings are displayable, you can configure the parameters in the following table.
SIP Header Warnings
Parameter Functiontemplate > parameter
Turn this feature on or off.sip-interop.cfg > voIpProt.SIP.header.warning.enable
Specify which warnings can be displayed.sip-interop.cfg > voIpProt.SIP.header.warning.codes.accept
Example Display of Warnings from SIP Headers Configuration
To enable the display of warnings from SIP headers, set the voIpProt.SIP.header.warning.enable
parameter in the features.cfg template to 1. Enter the warning codes as a comma-separated string. The
strings associated with the values 325 to 329 that display on the phone screen, as shown in the next
illustration, have been entered automatically by the call server and are not entered by the administrator in
the configuration file.
The following illustration shows a sample configuration from the sip-interop.cfg template file:
Set Up Server Redundancy
Server redundancy is often required in VoIP deployments to ensure continuity of phone service if, for
example, where the call server needs to be taken offline for maintenance, the server fails, or the connection
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between the phone and the server fails. The table Set Up Server Redundancy lists parameters you can
configure.
Two types of redundancy are possible:
● Failover—In this mode, full phone system functionality is preserved by having a second call server
of equivalent capability take over from the server that went down/off-line. Use this mode of operation
with DNS mechanisms or ‘IP Address Moving’ from the primary to the back-up server.
Caution: Old failover behavior is not supp orted
Prior to SIP 2.1, the reg.x.server.y parameters in <reg/> could be used for failover
configuration. The older behavior is no longer supported. Customers that are using the
reg.x.server.y.* configuration parameters where y>=2 should take care to ensure that their
current deployments are not adversely affected. For example, the phone only supports advanced SIP
features such as shared lines, missed calls, and presence with the primary server (y=1).
● Fallback—In this mode, a second call server of lesser capability (router or gateway device) takes
over call control to provide basic calling capability without some of the richer features offered by the
primary call server (for example, shared lines, presence, and message waiting indicator). Polycom
phones support configuration of multiple servers per SIP registration for this purpose.
In some cases, a combination of the two may be deployed. Consult your SIP server provider for
recommended methods of configuring phones and servers for failover configuration.
Note: Compatibility with Microsoft environments
The concurrent failover/fallback feature is not compatible with Microsoft environments.
Set Up Server Redundancy
Parameter Functiontemplate > parameter
Specify server redundancy options including failback mode,
failback timeout, and failover registration behavior.
Specify which server to contact if failover occurs.reg-advanced.cfg > reg.x.auth.optimizedInFailover
Override the default server redundancy options for a
specific registration.
.* indicates grouped parameters. See the section Example Two: Configuring Grouped Parameters for more
information.
Web Info: Failover configuration details
For more information, see Technical Bulletin 5844: SIP Server Fallback Enhancements on Polycom
Phones and Technical Bulletin 66546: Configuring Optional Re-Registration on Failover Behavior.
sip-interop.cfg > voIpProt.server.x.failOver.*
reg-advanced.cfg > reg.x.outboundProxy.failOver.*
Configure Presence
The presence feature enables you to monitor the status of other remote users and phones. By adding
remote users to your buddy list, you can monitor changes in the status of remote users in real time or you
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can monitor remote users as speed-dial contacts. You can also manually specify your status in order to
override or mask automatic status updates to others and you can receive notifications when the status of
your a remote line changes.
VVX phones support a maximum of 64 buddies for Open SIP server platforms and 200 contacts on Skype
for Business server. For information on the Skype for Business contacts, refer to the Polycom UC Software with Microsoft Lync Server and Skype for Business - Deployment Guide on Polycom Voice Support.
The following table lists the parameters you can configure. Note that other phone users can block you from
monitoring their phones.
For more information about the Microsoft Skype for Business or BroadSoft UC-One presence features, see
the Polycom VVX Business Media Phones User Guide.
Use the Presence Feature
Parameter Functiontemplate > parameter
Specify the line/registration number used to send SUBSCRIBE for
presence.
Specify if the MyStatus and Buddies soft keys display on the Home
screen.
Turn the presence feature on or off.features.cfg > feature.presence.enabled
features.cfg > pres.reg
features.cfg > pres.idleSoftkeys
Example Presence Configuration
In the following illustration, the presence feature has been enabled in feature.presence.enabled. The
MyStatus and Buddies soft keys both display on the phone’s home screen when you enable the
pres.idleSoftkeys parameter. The pres.reg parameter uses the address of phone line 1 for the
presence feature.
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This configuration enables the presence feature and display the MyStatus and Buddies soft keys on the
phone. When you press the Buddies soft key, contacts you have entered to your buddy list displays.
Provisional Polling of Polycom Phones
You can configure how your phone provisioning automatically by configuring the parameters in the table
Provisional Polling of Polycom Phones.
You can set the phone's automatic provisioning behavior to be:
● Absolute—The phone polls at the same time every day.
● Relative—The phone polls every x seconds, where x is a number greater than 3600.
● Random—The phone polls randomly based on a time interval you set.
If the time period is less than or equal to one day, the first poll is at a random time, x, between the
phone starting up and the polling period. Afterwards, the phone polls every x seconds.
If you set the polling period to be greater than one day with the period rounded up to the nearest
day, the phone polls on a random day based on the phone’s MAC address, and within a random
time set by the start and end polling time.
For example:
● If prov.polling.mode is set to rel and prov.polling.period is set to 7200, the phone polls
every two hours.
● If prov.polling.mode is set to abs and prov.polling.timeRandomEnd is set to 04:00, the
phone polls at 4am every day.
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● If prov.polling.mode is set to random, prov.polling.period is set to 604800 (7 days),
prov.polling.time is set to 01:00, prov.polling.timeRandomEnd is set to 05:00, and you
have 25 phones, a random subset of those 25 phones, as determined by the MAC address, polls
randomly between 1am and 5am every day.
● If prov.polling.mode is set to abs and prov.polling.period is set to 2328000, the phone
polls every 20 days.
Provisional Polling of Polycom Phones
Parameter Functiontemplate > parameter
To enable polling and set the mode, period, time, and time end parameters.site.cfg > prov.polling.*
.* indicates grouped parameters. See the section Example Two: Configuring Grouped Parameters for more
information.
Example Provisional Polling Configuration
The following illustration shows the default sample random mode configuration for the provisional polling
feature in the site.cfg template file. In this setup, every phone polls once per day, between 1 and 5 am.
Note: Only provision files when polling
If prov.startupCheck.enabled=’0’ then Polycom phones do not look for the sip.ld or the
configuration files when they are rebooted, lose power, or restarted. Instead, they look only when
receiving a checksync message, a polling trigger, or a manually started update from the menu or web
UI.
Some files such as bitmaps, .wav, the local directory, and any custom ringtones are downloaded each
time as they are stored in RAM and lost with every reboot.
Configure SIP Subscription Timers
This feature enables you to configure a subscription expiry independently of the registration expiry. You can
also configure an overlap period for subscription independently of the overlap period for the registration, and
a subscription expiry and subscription overlap for global SIP servers and per-registration SIP servers. Note
that per-registration configuration parameters override global parameters. If you have not explicitly
configured values for any user features, the default subscription values are used.
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SIP Subscription Timers
Parameter Functiontemplate > parameter
A global parameter that sets the phone’s requested subscription
period.
A global parameter that sets the number of seconds before the
expiration time returned by server x after which the phone
attempts to resubscribe.
A per-registration parameter that sets the phone’s requested
subscription period.
A per-registration parameter that sets the number of seconds
before the expiration time returned by server x after which the
phone attempts to resubscribe.
This section lists phone features for which no configuration is required.
Call Timer
A call timer displays on the phone’s screen. A separate call duration timer displays the hours, minutes, and
seconds of each call in progress.
Called Party Identification
By default, the phone displays and logs the identity of parties called from the phone. The phone obtains
called party identity from the network signaling. Because called party identification is a default state, the
phone displays caller IDs matched to the call server and does not match IDs to entries in the local contact
directory or corporate directory.
Connected Party Identification
By default, the phone displays and logs the identity of remote parties you connect to if the call server can
derive the name and ID from the network signaling. Note that in cases where remote parties have set up
certain call features, the remote party you connect to—and the caller ID that displays on the phone—may
be different than the intended party. For example, Bob places a call to Alice, but Alice has call diversion
configured to divert Bob’s incoming calls to Fred. In this case, the phone logs and displays the connection
between Bob and Fred. Note that the phone does not match party IDs to entries in the contact directory or
the corporate directory.
Microphone Mute
All phones have a microphone mute button. When you activate microphone mute, a red LED glows or a
mute icon displays on the phone screen, depending on the phone model you are using.
No configuration changes can be made to the microphone mute feature.
Automatic Gain Control
Automatic Gain Control (AGC) is applicable to conference phone models and is used to boost the transmit
gain of the local talker in certain circumstances. This increases the effective user-phone radius and helps
you to hear all participants equally.
Background Noise Suppression
Background noise suppression is designed primarily for handsfree operation and reduces background
noise, such as from fans, projectors, or air conditioners, to enhance communication.
Synthesized Comfort Noise Fill
This feature is an integral part of handsfree echo reduction; it is unrelated to Comfort Noise packets
generated if Voice Activity Detection is enabled. Synthesized Comfort Noise fill is designed to help provide
a consistent noise level to the remote user of a handsfree call. Fluctuations in perceived background noise
levels are an undesirable side effect of the non-linear component of most AEC systems. This feature uses
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noise synthesis techniques to smooth out the noise level in the direction toward the remote user, providing
a more natural call experience.
Jitter Buffer and Packet Error Concealment
The phone employs a high-performance jitter buffer and packet error concealment system designed to
mitigate packet inter-arrival jitter and out-of-order, or lost or delayed (by the network) packets. The jitter
buffer is adaptive and configurable for different network environments. When packets are lost, a
concealment algorithm minimizes the resulting negative audio consequences.
Configure Phone Alerts
This section shows you how to configure phone and call alert features.
Enable Persistent Mute
This feature enables you to have the mute state of your phone persist across calls. Default mute behavior
allows you to activate the mute state only if the phone is in an active call and ends when the active call ends.
When you enable this feature and press Mute, the phone stays in the mute state until you press Mute again
or until the phone restarts. When you mute the phone in an idle state, the mute LED glows but no icon
displays on the screen. When you initiate a new active call with mute on, the mute LED glows and a Mute
icon displays on the phone screen.
Persistent Mute Parameters
Parameter Functiontemplate > parameterEnable or disable the persistent mute feature.features.cfg > feature.persistentMute.enabled
Configure the Headset and Speakerphone
All VVX phones come with a handset and a dedicated connector for a headset and include support for a
USB headset; all Polycom phones have built-in speakerphones. You can enable and disable each of these
options, as shown in the following table. Note that although handsets are shipped with your phones,
headsets are not provided.
VVX phones have a dedicated key to switch between speakerphone and headset. You can enable or disable
the handsfree speakerphone mode.
Web Info: Configuring an external electronic hookswitch
You can configure all supported Polycom desktop phones with an external electronic hookswitch. For
more information, see Technical Bulletin 35150: Using an Electronic Hookswitch with SoundPoint IP
and Polycom VVX 1500 Phones.
Configure the Headset and Speakerphone
Parameter Functiontemplate > parameter
Enable or disable headset memory mode.reg-advanced.cfg and site.cfg > up.headsetMode
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Configure the Headset and Speakerphone (continued)
Enable or disable handsfree speakerphone mode.reg-advanced.cfg and site.cfg > up.handsfreeMode
Specify if the electronic hookswitch is enabled and
what type of headset is attached.
Specify if the handset or a headset should be used
for audio.
Specify how phone and the USB headset interact.site.cfg >up.headset.phoneVolumeControl
Specify if the USB headset volume persists
between calls.
reg-advanced.cfg and site.cfg > up.analogHeadsetOption
reg-advanced.cfg and site.cfg > up.audioMode
site.cfg > voice.volume.persist.headset
Example Handset, Headset, and Speakerphone Configuration
The following illustration shows the default settings in the reg-advanced.cfg template. In this example,
handsfree mode is enabled and headset memory mode and electronic hookswitch are disabled.
Apply Distinctive Ringing
The distinctive ringing feature enables you to apply a distinctive ringtone to a registered line, a specific
contact, or type of call.
There are three ways to set distinctive ringing, and the following table shows you the parameters for each.
If you set up distinctive ringing using more than one of the following methods, the phone uses the highest
priority method.
● You can assign ringtones to specific contacts in the contact directory. For more information, see Apply
Distinctive Incoming Call Treatment. This option is first and highest in priority.
● You can select a ringtone for each registered line on the phone. Press the Menu key, and select
Settings > Basic > Ring Type. This option has the lowest priority.
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● You can use the voIpProt.SIP.alertInfo.x.value and
voIpProt.SIP.alertInfo.x.class parameters in the sip-interop.cfg template to map calls to
specific ringtones. The value you enter depends on the call server. This option requires server
support and is second in priority.
Note:Using the SIP Alert-Info header to delay autoanswer
If you set delay=0 in the SIP.alert-Info header, the phone immediately auto-answers incoming calls
without ringing. If you set delay=x where x=time in seconds, the phone rings for that duration of time
before auto-answering incoming calls.
Apply Distinctive Ringing
Parameter Functiontemplate > parameterMap the Alert-Info string in the SIP header to ringtones.sip-interop.cfg > voIpProt.SIP.alertInfo.x.class
sip-interop.cfg > voIpProt.SIP.alertInfo.x.value
Specify a ringtone for a specific registered line.reg-advanced.cfg > reg.x.ringType
Specify ringtones for contact directory entries.000000000000-directory~.xml
Example Distinctive Ringing Configuration
The following illustration shows that the ring type ringer2 has been applied to incoming calls to line 1.
For a list of all parameters and their corresponding ringtones, see Ringtone Pattern Names.
Configure Do Not Disturb
You can use the do not disturb (DND) feature to temporarily stop incoming calls. You can also turn off audio
alerts and receive visual call alerts only, or you can make your phone appear busy to incoming callers.
Incoming calls received while DND is turned on are logged as missed.
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DND can be enabled locally through the phone or through a server. The table Configure Do Not Disturb lists
parameters for both methods. The local DND feature is enabled by default, and you have the option of
disabling it. When local DND is enabled, you can turn DND on and off using the Do Not Disturb button on
the phone. Local DND can be configured only on a per-registration basis. If you want to forward calls while
DND is enabled, see Configure Call Forwarding.
Note: Using do not disturb on shared lines
A phone that has DND enabled and activated on a shared line visually alerts you to an incoming call,
but the phone does not ring.
If you want to enable server-based DND, you must enable the feature on both a registered phone and on
the server. The benefit of server-based DND is that if a phone has multiple registered lines, you can apply
DND to all line registrations on the phone; however, you cannot apply DND to individual registrations on a
phone that has multiple registered lines. Note that although server-based DND disables the local Call
Forward and DND features, if an incoming is not routed through the server, you still receive an audio alert.
Server-based DND behaves the same way as the pre-SIP 2.1 per-registration feature with the following
exceptions:
● You cannot enable server-based DND if the phone is configured as a shared line.
● If server-based DND is enabled but not turned on, and you press the DND key or select DND on the
phone’s Features menu, the “Do Not Disturb” message displays on the phone and incoming calls
continue to ring.
Configure Do Not Disturb
Parameter Functiontemplate > parameter
Enable or disable server-based DND.sip-interop.cfg >
voIpProt.SIP.serverFeatureControl.dnd
Enable or disable local DND behavior when
server-based enabled.
Specify whether, when DND is turned on, the phone
rejects incoming calls with a busy signal or gives you a
visual and no audio alert.
Enable DND as a per-registration feature or use it as a
global feature for all registrations.
In the following example, taken from the sip-interop.cfg template, server-based DND has been enabled in
serverFeatureControl.dnd, and rejectBusyOnDnd has been set to 1 so that when you turn on DND
on the phone, incoming callers receive a busy signal.
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Note: DND LED alerts on the VVX
The LED on the Do Not Disturb key on the VVX 1500 is red when pressed or when server-based DND
is enabled.
Configure Call Waiting Alerts
By default, the phone alerts you to incoming calls while you are in an active call. As shown in the table
Configuring Call Waiting Alerts, you can disable call waiting alerts and you can specify the ringtone of
incoming calls.
Configuring Call Waiting Alerts
Parameter Functiontemplate > parameter
Enable or disable call waiting.sip-interop.cfg > call.callWaiting.enable
Specify the ringtone of incoming calls when you are in an active call.sip-interop.cfg > call.callWaiting.ring
Example Call Waiting Configuration
The following illustration shows you where to disable call waiting alerts and how to change the ringtone of
incoming calls in the sip-interop.cfg template.
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Configure Calling Party Identification
By default, the phone displays the identity of incoming callers if available to the phone through the network
signal. If the incoming call address has been assigned to the contact directory, you can choose to display
the name you assigned there. Note that the phone cannot match the identity of calling parties to entries in
the corporate directory.
Configure Calling Party Identification
Parameter Functiontemplate > parameter
Substitute the network address ID with the Contact Directory name.reg-advanced.cfg > up.useDirectoryNames
Override the default number of calls per line key for a specific line.reg-advanced.cfg > reg.x.callsPerLineKey
Example Calling Party Configuration
The following illustration shows you how to substitute the network address caller ID with the name you
assigned to that contact in the contact directory. The ID of incoming call parties displays on the phone
screen.
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Enable Missed Call Notification
You can display on the phone’s screen a counter that shows the number of missed calls. To reset the
counter, view the Missed Calls list on the phone. As the following table indicates, you can also configure the
phone to record all missed calls or to display only missed calls that arrive through the SIP server. You can
enable missed call notification for each registered line on a phone.
Enable Missed Call Notification
Parameter Functiontemplate > parameter
Enable or disable the missed call counter for a specific
registration.
Specify, on a per-registration basis, whether to display
all missed calls or only server-generated missed calls.
In the following example, the missed call counter is enabled by default for registered lines 1 and 2, and only
server-generated missed calls display on line 1.
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Apply Distinctive Incoming Call Treatment
You can apply distinctive treatment to specific calls and contacts in your contact directory. You can set up
distinctive treatment for each of your contacts by specifying a Divert Contact, enabling Auto-Reject, or by
enabling Auto-Divert for a specific contact in the local contact directory (see Use the Local Contact
Directory). You can also apply distinctive treatment to calls and contacts through the phone’s user interface.
Example Call Treatment Configuration
In the following example, the auto divert feature has been enabled in ad so that incoming calls from John
Doe are diverted to SIP address 3339951954 as specified in dc. Incoming calls from Bill Smith have been
set to auto reject in ar and are sent to voicemail.
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Note that if you enable both the auto divert and auto reject features, auto divert has precedence over auto
reject. For a list of all parameters you can use in the contact directory, see the table Understanding the Local
Contact Directory.
Apply Distinctive Call Waiting
You can use the alert-info values and class fields in the SIP header to map calls to distinct call-waiting types.
You can apply three call waiting types: beep, ring, and silent. The following table shows you the parameters
you can configure for this feature. This feature requires call server support.
Apply Distinctive Call Waiting
Parameter Functiontemplate > parameter
Enter the string which displays in the SIP Alert-Info header.sip-interop.cg > voIpProt.SIP.alertInfo.x.value
Enter the ring class name.sip-interop.cfg > voIpProt.SIP.alertInfo.x.class
Example Distinctive Call Waiting Configuration
In the following illustration, voIpProt.SIP.alertInfo.1.value is set to http://<SIP headerinfo>. An
incoming call with this value in the SIP Alert-Info header cause the phone to ring in a manner specified by
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voIpProt.SIP.alertInfo.x.class. In this example, the phone displays a visual LED notification, as
specified by the value visual.
Synthesized Call Progress Tones
Polycom phones play call signals and alerts, called call progress tones, such as busy signals, ringback
sounds, and call waiting tones. The built-in call progress tones on your phone match standard North
American tones. If you would like to customize the phone’s call progress tones to match the standard tones
in your region, contact Polycom Support.
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Configure Devices and Call Controls
Configure the Phone Display
This section provides information on setting up features available on the phone display screen.
Set the Time and Date Display
A clock and calendar are enabled by default. You can display the time and date for your time zone in several
formats, or you can turn it off altogether. You can also set the time and date format to display differently when
the phone is in certain modes. For example, the display format can change when the phone goes from idle
mode to an active call. You have to synchronize the phone to the Simple Network Time Protocol (SNTP)
time server. Until a successful SNTP response is received, the phone continuously flashes the time and date
to indicate that they are not accurate.
The time and date display on phones in PSTN mode are set by an incoming call with a supported caller ID
standard, or when the phone is connected to Ethernet and you enable the turn on the date and time display.
See the following table for basic time and display parameters.
Set the Time and Date Display
Parameter Functiontemplate > parameter
Turn the time and date display on or off.reg-advanced.cfg and site.cfg > up.localClockEnabled
Set the time and date display format.site.cfg > lcl.datetime.date.*
Display time in the 24-hour format.site.cfg > lcl.datetime.time.24HourClock
Set the basic SNTP settings and daylight savings
parameters.
.* indicates grouped parameters. See the section Example Two: Configuring Grouped Parameters for more
information.
site.cfg > tcpIpApp.sntp.*
Example Configuration
The following illustration shows an example configuration for the time and date display format. In this
illustration, the date is set to display over the time and in long format. The D, Md indicates the order of the
date display, in this case, day of the week, month, and day. In this example, the default time format is used,
or you can enable the 24-hour time display format.
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Use the table Date Formats to choose values for the lcl.datetime.date.format and lcl.datetime.date.longformat parameters. The table shows values for Friday, August 19, 2011.
Date Formats
lcl.datetime.date.formatlcl.datetime.date.longformatDate Displayed on Phone
dM,D019 Aug, Fri
dM,D119 August, Friday
Md,D0Aug 19, Fri
Md,D1August 19, Friday
D,dM0Fri, 19 Aug
D,dM1Friday, August 19
DD/MM/YYn/a19/08/11
DD/MM/YYYYn/a19/08/2011
MM/DD/YYn/a08/19/11
MM/DD/YYYYn/a08/19/2011
YY/MM/DDn/a11/08/19
YYYY/MM/DDn/a2011/08/11
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