Polycom Version 1.4.x 17, SoundPoint IP 300, SoundPoint IP 500, SoundPoint IP 600, SoundStation IP 4000 Administrator's Manual

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Administrator Guide

SoundPoint®/SoundStation® IP SIP

Version 1.4.x

17 November 2004

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Notices

1. Specifications subject to change without notice.

Polycom, Inc.

1565 Barber Lane, Milpitas CA 95035, USA

www.polycom.com

Part Number: 1725-11518-140 Rev A

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Administrator Guide - SoundPoint® IP / SoundStation® IP Table of Contents

Table of Contents

1 Overview ......................................................... 1

2 Installation and Operation ................................. 3

2.1 Installation Models ..................................................3

2.2 Installation Process..................................................4

2.2.1 Basic Network Setup....................................................................................................................... 4

2.2.1.1 DHCP or Manual TCP/IP Setup..............................................................................................5

2.2.1.2 Local User Interface Setup Menus ..........................................................................................6

2.2.1.3 Reset to Factory Defaults......................................................................................................9

2.2.2 Application Configuration................................................................................................................ 9

2.2.2.1 Centralized Configuration......................................................................................................10

2.2.2.2 Local Phone Configuration.....................................................................................................15

3 Features .......................................................... 17

3.1 Basic Features.........................................................17

3.1.1 Call Log ......................................................................................................................................... 17

3.1.2 Call Timer ...................................................................................................................................... 17

3.1.3 Call Waiting ................................................................................................................................... 18

3.1.4 Called Party Identification............................................................................................................... 18

3.1.5 Calling Party Identification.............................................................................................................. 18

3.1.6 Missed Call Notification................................................................................................................... 18

3.1.7 Configurable Feature Keys.............................................................................................................. 19

3.1.8 Connected Party Identification ........................................................................................................ 23

3.1.9 Context Sensitive Volume Control.................................................................................................... 23

3.1.10 Customizable Audio Sound Effects................................................................................................. 23

3.1.11 Message Waiting Indication .......................................................................................................... 24

3.1.12 Distinctive Incoming Call Treatment............................................................................................... 24

3.1.13 Distinctive Ringing........................................................................................................................ 25

3.1.13 Distinctive Call Waiting ................................................................................................................. 25

3.1.14 Do-Not-Disturb............................................................................................................................. 26

3.1.15 Handset, Headset, and Speakerphone........................................................................................... 26

3.1.16 Local Contact Directory................................................................................................................. 27

3.1.16.1 Local Contact Directory File Format.......................................................................................28

3.1.17 Local Digit Map............................................................................................................................ 30

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Administrator Guide - SoundPoint® IP / SoundStation® IP Table of Contents

3.1.18 Microphone Mute ......................................................................................................................... 30

3.1.19 Multiple Call Appearances............................................................................................................. 30

3.1.20 Shared Call Appearances .............................................................................................................. 31

3.1.21 Bridged Line Appearances............................................................................................................. 32

3.1.22 Customizable Fonts and Indicators................................................................................................. 33

3.1.23 Soft Key-Driven User Interface...................................................................................................... 34

3.1.24 Speed Dial ................................................................................................................................... 34

3.1.25 Time and Date Display.................................................................................................................. 35

3.1.26 Idle Display Animation ................................................................................................................. 36

3.2 Call Management Features........................................37

3.2.1 Automatic Off-hook Call Placement................................................................................................. 37

3.2.2 Call Hold........................................................................................................................................ 37

3.2.3 Call Transfer .................................................................................................................................. 38

3.2.4 Three-Way Conference, Local or Centralized.................................................................................... 38

3.2.5 Call Diversion (Call Forward) .......................................................................................................... 38

3.2.6 Directed Call Pick-up ...................................................................................................................... 39

3.2.7 Group Call Pick-up.......................................................................................................................... 40

3.2.8 Call Park / Retrieve ....................................................................................................................... 40

3.2.9 Last Call Return.............................................................................................................................. 41

3.3 Audio Processing Features ........................................41

3.3.1 Low-Delay Audio Packet Transmission............................................................................................. 41

3.3.2 Jitter Buffer and Packet Error Concealment ..................................................................................... 41

3.3.3 Local Conference Mixing................................................................................................................. 42

3.3.4 Voice Activity Detection (VAD)......................................................................................................... 42

3.3.5 DTMF Tone Generation ................................................................................................................... 43

3.3.6 DTMF Event RTP Payload ................................................................................................................ 43

3.3.7 Acoustic Echo Cancellation (AEC) ..................................................................................................... 44

3.3.8 Audio Codecs.................................................................................................................................. 44

3.3.9 Background Noise Suppression (BNS).............................................................................................. 45

3.3.10 Comfort Noise Fill......................................................................................................................... 45

3.3.11 Automatic Gain Control (AGC)....................................................................................................... 45

3.4 Presence and Instant Messaging Features ...................45

3.4.1 Presence........................................................................................................................................ 45

3.4.2 Instant Messaging .......................................................................................................................... 46

3.5 Localization Features ...............................................46

3.5.1 Multilingual User Interface ............................................................................................................. 46

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3.5.2 Downloadable Fonts ....................................................................................................................... 48

3.5.3 Synthesized Call Progress Tones...................................................................................................... 48

3.6 Advanced Server Features ........................................49

3.6.1 Voicemail Integration ..................................................................................................................... 49

3.6.2 Multiple Registrations..................................................................................................................... 50

3.6.3 ACD login / logout ......................................................................................................................... 52

3.6.4 ACD agent available / unavailable.................................................................................................. 53

3.6.5 Server Redundancy ........................................................................................................................ 53

3.6.5.1 DNS SIP Server Name Resolution...........................................................................................54

3.7 Accessory Internet Features.......................................54

3.7.1 MicroBrowser................................................................................................................................. 54

3.8 Security Features.....................................................54

3.8.1 Local User and Administrator Privilege Levels.................................................................................. 54

3.8.2 Incoming Signaling Validation......................................................................................................... 55

4 Optimization .................................................... 57

4.1 Ethernet Switch .......................................................57

4.2 Application Network Setup .......................................57

4.2.1 RTP Ports....................................................................................................................................... 57

4.2.2 Working with Network Address Translation (NAT) ............................................................................58

4.3 Updating and Rebooting...........................................59

4.4 Event Logging .........................................................60

4.5 Audio Quality Issues and VLANs ................................61

4.5.1 IP TOS ........................................................................................................................................... 62

4.5.2 IEEE 802.1p/Q............................................................................................................................... 62

4.5.3 RTCP Support ................................................................................................................................. 63

4.6 Configuration Files...................................................65

4.6.1 Core Configuration - ipmid.cfg ........................................................................................................ 65

4.6.1.1 Localization <localization/> .................................................................................................65

4.6.1.2 User Preferences <user_preferences/> ..................................................................................69

4.6.1.3 Tones <tones/> .................................................................................................................70

4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> ..............................................................72

4.6.1.5 Sound Effects <sound_effects/>...........................................................................................73

4.6.1.6 Voice Settings <voice/> ......................................................................................................78

4.6.1.7 Quality of Service <QOS/> ..................................................................................................86

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4.6.1.8 Basic TCP/IP <TCP_IP/>.....................................................................................................88

4.6.1.9 Web Server <HTTPD/>........................................................................................................91

4.6.1.10 Call Handling Configuration <call/>.....................................................................................92

4.6.1.11 Directory <directory/>.......................................................................................................93

4.6.1.12 Fonts <font/>..................................................................................................................94

4.6.1.13 Keys <keys/>..................................................................................................................96

4.6.1.14 Bitmaps <bitmaps/>.........................................................................................................100

4.6.1.15 Indicators <indicators/>.....................................................................................................100

4.6.1.16 Event Logging <logging/> .................................................................................................103

4.6.1.17 Security <security/> .........................................................................................................106

4.6.1.18 Provisioning <provisioning/>..............................................................................................107

4.6.1.19 RAM Disk <RAMdisk/>......................................................................................................107

4.6.1.20 Request <request/>..........................................................................................................108

4.6.1.21 Feature <feature/>...........................................................................................................109

4.6.1.22 Resource <resource/>.......................................................................................................110

4.6.1.23 MicroBrowser <microbrowser/>..........................................................................................111

4.6.2 SIP Configuration - sip.cfg .............................................................................................................. 113

4.6.2.1 Protocol <volpProt/>...........................................................................................................113

4.6.3 Per-phone Configuration - phone1.cfg............................................................................................. 121

4.6.3.1 Registration <reg/> ............................................................................................................121

4.6.3.2 Calls <call/>......................................................................................................................123

4.6.3.3 Diversion <divert/>.............................................................................................................124

4.6.3.4 Dial Plan <dialplan/>..........................................................................................................126

4.6.3.5 Messaging <msg/> ............................................................................................................128

4.6.3.6 Network Address Translation <nat/> .....................................................................................129

5 Session Initiation Protocol (SIP)........................... 131

5.1 Basic Protocols ........................................................131

5.1.1 RFC and Internet Draft Support....................................................................................................... 131

5.1.2 Request Support............................................................................................................................. 131

5.1.3 Header Support.............................................................................................................................. 132

5.1.4 Response Support........................................................................................................................... 134

5.1.4.1 1xx Responses - Provisional ..................................................................................................134

5.1.4.2 2xx Responses - Success ......................................................................................................134

5.1.4.3 3xx Responses - Redirection..................................................................................................135

5.1.4.4 4xx Responses - Request Failure............................................................................................135

5.1.4.5 5xx Responses - Server Failure..............................................................................................136

5.1.4.6 6xx Responses - Global Failure ..............................................................................................136

5.1.5 Hold Implementation...................................................................................................................... 137

5.1.6 Reliability of Provisional Responses................................................................................................. 137

5.1.7 Transfer......................................................................................................................................... 137

5.1.8 Third Party Call Control................................................................................................................... 137

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5.2 Protocol Extensions..................................................138

5.2.1 RFC and Internet Draft Support....................................................................................................... 138

5.2.2 Request Support............................................................................................................................. 138

5.2.3 SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE).......................................... 139

5.2.4 Shared Call Appearance Signaling................................................................................................... 139

5.2.5 Bridged Line Appearance Signaling ................................................................................................. 139

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Administrator Guide - SoundPoint® IP / SoundStation® IP Table of Contents

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Administrator Guide - SoundPoint® IP / SoundStation® IP Overview

1 Overview

This Administrator Guide is for the SIP 1.4.0 software release, and the bootROM 2.6.0 release.

SoundPoint nications terminals for Ethernet TCP/IP networks. They are designed to facilitate high-quality audio and text message communications. These phones are endpoints in the overall network topology designed to interoperate with other compatible equipment including application servers, media servers, internetworking gateways, voice bridges, and other endpoints.

IP and SoundStation® IP are feature-rich, enterprise-class voice commu-

Remote

Boot Server

10/100

Ethernet

Switch

Ethernet

Internet

789101112

123456

789101112

123456

Router / Firewall

12x6x8x2x9x3x10x4x11x5x7x

Polycom

SoundPoint IP

500/600s

10/100

12x6x8x2x9x3x10x4x11x5x7x

Ethernet

Hub

IDC

789101112

Ethernet

123456

789101112

Ethernet

123456

Remote

Application

Server

12x6x8x2x9x3x10x4x11x5x7x

PSTN

Gateway

Modem Bank

12x6x8x2x9x3x10x4x11x5x7x

Ethernet

12x6x8x2x9x3x10x4x11x5x7x

Switches

Voice Bridge

IDC

Local

Application

Server

Local

Boot Server

PC PC

The phones connect physically to a standard office twisted-pair (IEEE 802.3) 10/100 megabytes per second Ethernet LAN and send and receive all data using the same packet-based technology. Since the phone is a data terminal, digitized audio being just

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Administrator Guide - SoundPoint® IP / SoundStation® IP Overview

another type of data from its perspective, the phone is capable of vastly more than tra-

ditional business phones. As SoundPoint

IP and SoundStation® IP run the same protocols as your office personal computer, many innovative applications can be developed without resorting to specialized technology. Regardless of the diverse application potential, it is fundamentally a good office phone, providing the productivity enhancing features needed today such as multiple call appearances, full-duplex speakerphone, hold, transfer, conference, forward, voice mail compatibility, and contact directory.

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Administrator Guide - SoundPoint® IP / SoundStation® IP Installation and Operation

2 Installation and Operation

This section describes the basic steps that are needed to make your phone operational.

2.1 Installation Models

There are diverse installation models scaling from stand-alone phones to large, centrally provisioned systems with thousands of phones. For any size system, the phones can be centrally provisioned from a boot server via a system of global and per-phone configuration files. To augment the central provisioning model, or as the sole method in smaller systems, configuration can be done using user interfaces driven from the phones themselves: both a local setup user interface and a web server-based user interface are available to make configuration changes.

master config file

application binary

config files

dictionary files

user interface resource files

Boot Server

0004f2002999.cfg sip.ld ipmid.cfg sip.cfg phone2999.cfg SoundPointIP-dictionary.xml

Local User Interface

0004f2002999-phone.cfg 0004f2002999-directory.cfg

SoundPoint IP

MAC 00:04:f2:00:29:99

config overrides directory

0004f2002999-boot.log 0004f2002999-app.log

Local

Web Server

event log files

A boot server allows global and per-phone configuration to be managed centrally via text XML-format configuration files that are downloaded by the phones at boot time. The boot server also facilitates automated application upgrades, diagnostics, and a measure of fault tolerance.

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The configuration served by the boot server can be augmented by changes made locally on the phone itself or via the phone’s built-in web server. If file uploads are permitted, the boot server allows these local changes to be backed up automatically.

Polycom recommends the boot server central provisioning model for installations involving more than a few phones. The investment required is minimal in terms of time and equipment, and the benefits are significant.

The phones also support dynamic host configuration protocol (DHCP). When set up, DHCP permits plug-and-play TCP/IP network setup.

2.2 Installation Process

Regardless of whether or not you will be installing a centrally provisioned system, there are two steps required to get your phones up and running.

1. Basic TCP/IP Network Setup such as IP address and subnet mask. For more information, see 2.2.1 Basic Network Setup on page 4.

2. Application Configuration such as application specific parameters. For more information, see

2.2.2 Application Configuration on page 9.

2.2.1 Basic Network Setup

The phones boot up in two phases:

• Phase 1: bootROM - a generic program designed to load the application.

• Phase 2: application - the SIP phone application.

Networking starts in Phase 1. The bootROM application uses the network to query the boot server for upgrades or configuration changes, which is an optional process that will happen automatically when properly deployed. The boot server can be on the local LAN or anywhere on the Internet. The bootROM then loads the configured application. The application will restart networking using most of the parameters established by the bootROM (a DHCP query will be performed by the application).

Basic network settings can be changed during Phase 1 using the bootROM’s setup menu. A similar, but more sophisticated menu system is present in the application for changing the same network parameters. For more information, see 2.2.1.2 Local User Interface Setup Menus on page 6.

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Administrator Guide - SoundPoint® IP / SoundStation® IP Installation and Operation

2.2.1.1 DHCP or Manual TCP/IP Setup

Basic network settings can be derived from DHCP or entered manually using the phone’s LCD-based user interface. Polycom recommends using DHCP where possible to eliminate repetitive manual data entry.

The following table shows the manually entered networking parameters that may be overridden by parameters obtained from a DHCP server:

Parameter DHCP Option DHCP

12 3

IP address

subnet mask

IP gateway

boot server address

SNTP server address

SNTP GMT offset

DNS server IP address

alternate DNS server IP

See 2.2.1.2.2 DHCP Menu

on page 7

42 then 4

•- •

•• •

•- •

address

DNS domain

•- •

Configuration File (Phase 2: application only)

 priority when more than one source exists 

Local FLASH

VLAN ID

a. Can be obtained from a connected Ethernet switch if the switch supports CDP.

See 2.2.1.2.2 DHCP Menu

on page 7

Special Case: Cisco Discovery Protocol (CDP)

rides Local FLASH which overrides DHCP VLAN

Discovery.

over-

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Administrator Guide - SoundPoint® IP / SoundStation® IP Installation and Operation

2.2.1.2 Local User Interface Setup Menus

Access to Network Configuration Menu:

Phase 1: bootROM The network configuration menu is accessible during the auto-boot

countdown of the bootROM phase of operation. Press the key to launch the main menu.

Phase 2: application The network configuration menu is accessible from the main menu.

Navigate to Menu>Settings>Network Configuration. This menu is locked by default. Enter the administrator password to unlock. (Factory default password: 456)

Phone network configuration parameters may be edited by means of a main menu and two sub-menus: DHCP Menu and Server Menu.

Setup soft

Use the soft keys, the arrow keys, the Sel/

Parameters that cannot be changed are read-only due to the value of other parameters. For example, if the DHCP Client parameter is enabled, the Phone IP Addr and Subnet Mask parameters are dimmed or not visible since these are guaranteed to be supplied by the DHCP server (mandatory DHCP parameters) and the statically assigned IP address and subnet mask will never be used in this configuration.

2.2.1.2.1 Main Menu

Configuration parameters that may be edited on the main setup menu are described in the table below:

Name Possible Values

DHCP Client Enabled, Disabled If enabled, DHCP will be used to obtain the

Phone IP Address dotted-decimal IP

address

, and the Del/X keys to make changes.

Description

parameters discussed in 2.2.1.1 DHCP or Manual TCP/IP Setup on page 5.

Phone’s IP address.

Note: Disabled when DHCP client is enabled.

Subnet Mask dotted-decimal subnet

IP Gateway dotted-decimal IP

SNTP Address dotted-decimal IP

mask

address

domain name string

Phone’s subnet mask.

Note: Disabled when DHCP client is enabled.

Phone’s default router.

SNTP server from which the phone will obtain the current time.

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Administrator Guide - SoundPoint® IP / SoundStation® IP Installation and Operation

Name Possible Values

GMT Offset -12 through +13 Offset of the local time zone from Greenwich

DNS Server dotted-decimal IP

DNS Alternate Server dotted-decimal IP

DNS Domain domain name string Phone’s DNS domain.

CDP Enabled, Disabled If enabled, the phone will attempt to determine

VLAN ID Null, 0 through 4095 Phone’s 802.1Q VLAN identifier.

a. A parameter value of “???” indicates that the parameter has not yet been set and saved in the

phone’s configuration. Any such parameter should have its value set before continuing.

The DHCP and Server sub-menus may be accessed from the main setup menu.

2.2.1.2.2 DHCP Menu

address

Description

Mean Time in half hour increments.

Primary server to which the phone directs Domain Name System queries.

Secondary server to which the phone directs Domain Name System queries.

its VLAN ID via the CDP.

Note: 4095 = no VLAN tagging

The DHCP menu is accessible only when the DHCP client is enabled. DHCP configuration parameters are described in the following table:

Possible

Name

Timeout 1 through 600 Number of seconds the phone waits for secondary

Boot Server Option 66

Values

Custom

Static

Custom+Opt.66

Description

DHCP Offer messages before selecting an offer.

Option 66: The phone will look for option number 66 (string type) in the response received from the DHCP server. The DHCP server must be configured to send the boot server address in option 66.

Custom: The phone will look 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 will use the boot server configured via the Server Menu. For more information, see

2.2.1.2.3 Server Menu on page 8.

Custom+Opt.66: The phone will first use the custom option if present or use Option 66 if the custom option is not present.

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Possible

Name

Values

Description

Boot Server Option 128 through 254

Boot Server Option Type

VLAN Discovery Disabled No VLAN discovery via DHCP.

VLAN ID Option 128 through 254

2.2.1.2.3 Server Menu

When the boot server parameter is set to Custom, this (Cannot be the same as VLAN

parameter specifies the DHCP option number in which

the phone will look for its boot server. ID Option)

IP Address, String

When the Boot Server parameter is set to Custom, this

parameter specifies the type of the DHCP option in

which the phone will look for its boot server.

Fixed Use predefined DHCP private option values of 128,

144, 157 and 191. If this is used, the VLAN ID Option

field will be ignored.

Custom Use the number specified in the VLAN ID Option field

as the DHCP private option value.

The DHCP private option value (when VLAN Discov(Cannot be the

ery is set to Custom). Default is 129. same as Boot Server Option)

Name Possible Values Description

Server Type

FTP, Trivial FTP

Server Address dotted-decimal IP address

domain name string

FTP User

FTP Password

any string When the Server Type parameter is set to FTP,

a. Using TFTP will make management of the phone more difficult. For more information, see

2.2.2.1.1 FTP vs. TFTP on page 10.

When set to FTP, the phone will use the File Transfer Protocol (FTP) to obtain configuration and phone application files from the boot server.

When set to Trivial FTP, the phone will use the Trivial File Transfer Protocol (TFTP) to obtain configuration and phone application files from the boot server.

The boot server to use if the DHCP client is disabled, or the DHCP server does not send a boot server option, or the Boot Server parameter is set to Static.

this is the user name used when the phone logs into the FTP server.

this is the password used when the phone logs in to the FTP server.

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b. The FTP server user name and FTP password should be changed from the default values.

Note that FTP is an insecure protocol and the user chosen should have very few privileges on the server.

2.2.1.3 Reset to Factory Defaults

The basic network configuration referred to in the preceding sections can be reset to factory defaults. To perform this function on the IP 300, 500 and 600, simultaneously press and hold the 4, 6, 8 and * dial pad keys until the password prompt appears. To perform this function on the IP 4000, simultaneously press and hold the 6, 8 and * dial pad keys until the password prompt appears. Enter the administrator password to initiate the reset. This will reset the administrator password as well.

2.2.2 Application Configuration

While it is possible to make calls with the phone using its default configuration, most installations will require some basic configuration changes to get things running optimally. These changes can be made using the central boot server model, if a boot server has been set up, or some, but not all changes can be made using the phone’s internal configuration web server.

Advantages of using a boot server:

1. The centralized repository for application images and configuration files permits application updates and coordinated configuration parameters.

2. Some parameters can only be modified using boot server configuration files.

3. The multilingual feature requires boot server-resident dictionary files.

4. The customized sound effect wave files require a boot server.

5. When file uploads are permitted, the boot server is the repository for:

• boot process and application event log files - very effective when diagnosing system problems

• local configuration changes via the <Ethernet address>-phone.cfg boot server configuration overrides file - the phone treats the boot server copy as the original when booting

• per-phone contact directory named <Ethernet address>-directory.cfg

6. The boot server copy of the application images and configuration files can be used to “repair” a damaged phone configuration in the same way that system repair disks work for PCs.

The following sections discuss the available configuration options.

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Administrator Guide - SoundPoint® IP / SoundStation® IP Installation and Operation

2.2.2.1 Centralized Configuration

The phone application consists of an executable image file (sip.ld) and one or more XML-format configuration files. In the centrally provisioned model, these files are stored on a boot server and cached in the phone. If the boot server is available at boot time, the phone will automatically synchronize its configuration cache with the boot server: bootROM image, application executable, and configuration files are all upgraded this way. The phone requires that a SNTP server be properly configured for this feature to work optimally.

2.2.2.1.1 FTP vs. TFTP

Note

If the phone has bootROM version 2.6.0 or greater, the following restrictions do not apply.

If using a trivial FTP (TFTP) protocol boot server, synchronization with the boot server will be based on file names, not file timestamps, as is the case with FTP. Executable upgrades and changes made to configuration files will not be recognized by the phone unless their names are changed.

Example:

file.ld  file01.ld (an upgrade to SIP 1.0.1)

ipmid.cfg  ipmid01.cfg

The master configuration file, which references the other files and is always downloaded and parsed by the phone, will need to change to reflect these name changes.

Polycom does not recommend TFTP boot servers for actively managed systems. File name management is the responsibility of the System Administrator.

2.2.2.1.2 Configuration Files

The phone configuration files consist of master configuration files and application configuration files.

2.2.2.1.2.1 Master Configuration Files

Central provisioning requires that an XML-format master configuration file be located in the home directory on the boot server.

Per-phone Master Configuration File

If per-phone customization is required (for all applications that require per-phone customization), the file should be named <Ethernet address>.cfg, where Ethernet address

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Administrator Guide - SoundPoint® IP / SoundStation® IP Installation and Operation

is the Ethernet MAC address of the phone in question. For A-F hexadecimal digits, use lower case only, for example, 0004f200106c.cfg. The Ethernet address can be viewed using the

About soft key during the auto-boot countdown of the bootROM or

via the Menu>System Status>General menu in the application. It is also printed on a label on the back of the phone.

Default Master Configuration File

For systems in which the configuration is identical for all phones (no per-phone <Ethernet address>.cfg files), the default master configuration file may be used to set the configuration for all phones. The file named 000000000000.cfg (<12 zeros>.cfg) is the default master configuration file and it is recommended that one be present on the boot server. If a phone does not find its own <Ethernet address>.cfg file, it will use this one, and establish a baseline configuration. This file is part of the standard Polycom distribution of configuration files. It should be used as the template for the <Ethernet address>.cfg files.

The default SIP master configuration file, 000000000000.cfg, is shown below:

Example:

<?xm l version= "1 .0" stand alo ne ="ye s" ?>

Default Master SIP Configuration File -->

<!-<!--

Edit and rename this file to <Ethernet-address>.cfg for each

phone.

<!-- < APPLICATION APP_FILE_PATH="sip.ld"

-->

$Revision: 1.24 $ $Date: Mar 26 2003 11:59:02 $ -->

CONFIG_FILES="phone1.cfg, sip.cfg, ipmid.cfg" MISC_FILES= "" LOG_FILE_DIRECTORY=""/>

Master configuration files contain four XML attributes:

APP_FILE_PATH The path name of the application executable. Has a maximum length

of 127 characters.

CONFIG_FILES A comma-separated list of configuration files. Has a maximum length

of 511 characters, including commas and white space.

MISC_FILES

LOG_FILE_DIRECTORY An alternative directory to use for log files if required. This is left

a. MISC_FILES is not used for the SIP application, but is used for the MGCP application.

A comma-separated list of other required files.

blank by default.

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Note

The order of the configuration files listed in CONFIG_FILES is significant.

• The files are processed in the order listed (left to right).

• The same parameters may be included in more than one file.

• The parameter found first in the list of files will be the one that is effective. This provides a convenient means of overriding the behavior of one or more phones without altering the baseline configuration files for an entire system.

2.2.2.1.2.2 Application Configuration Files

Typically, the files are arranged in the following manner although parameters may be moved around within the files and the file names themselves can be changed as needed.

Per-phone settings  phoneXXXX.cfg

Application settings  sip.cfg

Core settings  ipmid.cfg

Category Description Example

Core Contain parameters that affect the basic operation of the phone

such as voice codecs, gains, and tones. All phones in an installation usually share this category of files.

Applicationspecific

User Contain parameters unique to a particular phone user. Typical

Contain parameters that dictate performance of a particular phone application.

Typical parameters include the IP address of an application server. All phones in an installation usually share application-specific files. This file would normally be modified from Polycom templates.

parameters include:

• display name

• unique addresses

Each phone in an installation usually has its own customized version of user files derived from Polycom templates.

ipmid.cfg

sip.cfg, mgcp.cfg

phone1.cfg

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These application configuration files dictate the behavior of the phone once it is running the executable specified in the master configuration file.

Important

Configuration files should only be modified by a knowledgeable System Administrator. Applying incorrect parameters may render the phone unusable.

2.2.2.1.3 Deploying a Boot Server for the Phones

The following table describes the steps required for successful deployment of a boot

server for SoundPoint

IP and SoundStation® IP phones. These instructions assume

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an FTP protocol boot server. For more information on TFTP boot servers, see

2.2.2.1.1 FTP vs. TFTP on page 10.

Step: Instructions:

1. Set-up FTP Server:

Install FTP server application or locate suitable existing server. Use RFC-compliant FTP servers. The fol-

Note: Typically all phones are configured with the same FTP account, but the FTP account provides a means of conveniently partitioning the configuration. Give each account an unique home directory on the server and change the configuration on an account-by-account basis.

lowing FTP servers have been tested with

SoundPoint

IP and SoundStation® IP and are known

to work acceptably:

• Linux: ProFTPd 1.2.2 through 1.2.9 rc2p, ftpdbsd-0.3.3 (Linux port), we-ftpd 2.6.0

• Windows 2000 Server: IIS 5.0, WFTPD 2.03

Create account and home directory.

2. Copy all files: Copy all files from the distribution.zip file to the phone home directory. Maintain the same folder hierarchy.

3. Create per-phone configuration

files

Obtain a list of phone Ethernet addresses (barcoded label on underside of phone).

Create per-phone phoneXXXX.cfg and <Ethernet address>.cfg files by using the 00000000000.cfg and phone1.cfg files from the distribution as templates.

Edit contents of phoneXXXX.cfg as appropriate. For example, edit the registration parameters.

Edit the CONFIG_FILES attribute of the <Ethernet

address>.cfg files so that it references the appropriate phoneXXXX.cfg file. (Replace the reference to

phone1.cfg with phoneXXXX.cfg.)

4. Edit <application>.cfg: See 4.6.2 SIP Configuration - sip.cfg on page 113, particularly for SIP server address.

5. Edit ipmid.cfg: See 4.6.1 Core Configuration - ipmid.cfg on page 65.

Most of the default settings are typically adequate, however, if overriding SNTP settings are not available via DHCP, the SNTP GMT offset and (possibly) the SNTP server address will need to be edited for the correct local conditions. Changing the default daylight savings parameters will likely be necessary outside of North American locations.

(Optional) Disable the local web (HTTP) server or alter its signalling port if local security policy dictates.

Change the default location settings:

• user interface language

• time and date format

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Step: Instructions:

6. Decide on FTP server security pol-

icy:

7. Reboot phones after configuring

their boot server via DHCP or statically:

Polycom recommends allowing file uploads to the FTP server where the security environment permits. This allows event log files to be uploaded and changes made by the phone user to the configuration (via the web server and local user interface) and changes made to the directory to be backed up. For organizational purposes, configuring a separate log file directory is recommended, but not required (see LOG_FILE_DIRECTORY in 2.2.2.1.2.1 Master Configuration Files on page 10). File permissions should give the minimum access required, and the account used should have no other rights on the server. The phone's FTP account needs to be able to add files to which it can write in the log file directory and the root directory. It must also be able to list files in all directories mentioned in the [mac].cfg file. All other files that the phone needs to read, such as the application executable and the standard configuration files, should be made read-only via file server file permissions.

See 2.2.1 Basic Network Setup on page 4.

To reboot phones manually, press and hold the following keys simultaneously until a confirmation tone is heard or for about three seconds:

IP 300: Volume-, Volume+, Hold and Redial

IP 500: Volume-, Volume+, Hold, and Messages

IP 600: Volume-, Volume+, Mute, and Messages

IP 4000: *, #, Volume+, and Select

Monitor the boot server event log and the uploaded event log files (if permitted):

Ensure that the configuration process completed correctly.

Start making calls!

a. The FTP account name and password must match those configured in the phones them-

selves. Defaults are: name: PlcmSpIp, password: PlcmSpIp

b. This step may be omitted if per-phone configuration is not needed.

2.2.2.2 Local Phone Configuration

As the only method of modifying phone configuration or as a distributed method of augmenting a centralized provisioning model, a local phone-based configuration web server is available, unless disabled via ipmid.cfg. For more information, see 4.6.1.9 Web Server <HTTPD/> on page 91. The phone’s local user interface also permits

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many application settings to be modified, such as SIP server address or ring type or regional settings such as time/date format and language.

Local Web Server Access Point your web browser to http://<phoneIPAddress>/.

Configuration pages are accessible from the menu along the top banner.

The web server will issue an authentication challenge to all pages except for the home page.

Credentials are (case sensitive):

• User Name: Polycom

• Password: The administrator password is used for this.

Local Settings Menu Access Some items in the Settings menu are locked to prevent accidental

changes. To unlock these menus, enter the user or administrator passwords.

The administrator password can be used anywhere that the user password is used.

Factory default passwords are:

• User password: 123

• Administrator password: 456

Passwords:

Network Configuration Requires administrator password.

Authentication Requires user password.

Reset User Settings Requires user password.

Changes made via the web server or local user interface are stored internally as overrides. These overrides take precedence over settings contained in the configuration obtained from the boot server that existed previously within the phone.

If the boot server permits uploads, these override setting will be saved in a file called <Ethernet address>-phone.cfg on the boot server.

Important

Local configuration changes will continue to override the boot server-derived configuration until deleted via the Reset User Settings menu selection.

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3 Features

This section describes the many features and corresponding administration points of

SoundPoint uration Files on page 65.

3.1 Basic Features

3.1.1 Call Log

The phone maintains a call log. The log:

• contains call information such as remote party identification, time and date, and call duration,

• allows for convenient redialing of previous outgoing calls and for returning incoming calls,

• can be used to save contact information from call log entries to the contact directory.

IP and SoundStation® IP. References are made frequently to 4.6 Config-

The call log is stored in volatile memory and is maintained automatically by the phone in three separate lists; Missed Calls, Received Calls and Placed Calls. The call lists can be cleared manually by the user and will be erased on reboot.

Central (boot server)

Local

Configuration File: ipmid.cfg

Web S e rver (if enabled)

Local Telephone User Interface

3.1.2 Call Timer

A call timer is provided on the display. A separate call timer is maintained for each distinct call in progress.

Enable or disable all call lists or individual call lists.

• For more information, see 4.6.1.21 Feature <feature/> on page 109.

None.

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3.1.3 Call Waiting

When an incoming call arrives while the user is active on another call, the incoming call is presented to the user visually on the LCD display. A configurable sound effect such as the familiar call-waiting beep will be mixed with the active call audio as well.

3.1.4 Called Party Identification

The phone displays and logs the identity of the remote party specified for outgoing calls. This is the party that the user intends to connect with.

3.1.5 Calling Party Identification

The phone displays the caller identity, derived from the network signalling, when an incoming call is presented. For calls from parties for which a directory entry exists, the local name assigned to the directory entry may optionally be substituted.

Central (boot server)

Local

Configuration File: ipmid.cfg

Web S e rver (if enabled)

Local Telephone User Interface

Specify whether or not to use directory name substitution.

• For more information, see 4.6.1.2 User Preferences <user_preferences/> on page 69.

Specify whether or not to use directory name substitution. Navigate to: http://<phoneIPAddress>/coreConf.htm#us

Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. Changes will permanently override global settings unless deleted via the Reset User Settings menu selection.

None.

3.1.6 Missed Call Notification

The phone can display the number of calls missed since the user last looked at the Missed Calls list. The types of calls which are counted as “missed” can be configured per registration. Remote missed-call notification can be used to notify the phone when a call originally destined for it is diverted by another entity such as a SIP server.

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Central (boot server)

Local

Configuration file: ipmid.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, see 4.6.1.21 Feature <feature/> on page 109.

Specify per-registration whether all missed-call events or only remote/server-generated missed-call events will be displayed.

• For more information, see 4.6.3.2.3 Missed Call Configuration <serverMissedCall/> on page 123.

None.

3.1.7 Configurable Feature Keys

All key functions can be changed from the factory defaults, although this is typically not necessary. The scrolling timeout for specific keys can be configured.

Central (boot server)

Local

Configuration File: ipmid.cfg

Web S erv e r (if enabled)

Local Telephone User Interface

Set the key scrolling timeout, key functions, and sub-pointers for each key (usually not necessary).

• For more information, see 4.6.1.13 Keys <keys/> on page 96.

None.

The following diagrams and table show the default SIP key layouts for

SoundPoint

IP 300, IP 500, IP 600 and SoundStation® IP 4000 models.

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SoundPoint IP 300 SIP Key Layout

Sel

28 27 25

2635

ABC21DEF

192021

DEF

ABC

MNOJKLGHI

16 17 18

GHI

PQRS

JKL

TUV87WXYZ

TUV

OPER

MNO

WXYZ

131415

121110

SoundPoint IP 500 SIP Key Layout

Conference

Directories

Services

Call Lists

Conference

Transfer

Redial

19 20 21

GHI

PQRS

ABC21DEF

ABC

JKL

TUV87WXYZ

TUV

OPER

222324

DEF

MNOJKLGHI

MNO

WXYZ

Del

Do Not Disturb

Redial

Hold

Key ID

Sel

Del

Messages

Do Not Disturb

Hold

Key ID

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SoundPoint IP 600 SIP Key Layout

Directories

Services

Conference

Transfer

Redial

Hold

ABC21DEF

ABC

19 20 21

JKL

GHI

PQRS

TUV87WXYZ

TUV

OPER

MNO

WXYZ

25262728

222324

DEF

MNOJKLGHI

131415

Key ID

SoundStation IP 4000 SIP Key Layout

Sel

Del

Messages

Do Not Disturb

Hold

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Key ID IP 300 Function IP 500 Function IP 600 Function IP 4000 Function

1 Line1 ArrowUp ArrowUp Dialpad1

2 Line2 ArrowLeft ArrowLeft Dialpad2

3 n/a Select ArrowDown Dialpad3

4 n/a ArrowRight ArrowRight VolUp

5 Hold ArrowDown Select Handsfree

6 n/a Delete Delete ArrowUp

7 DoNotDisturb Menu Menu Dialpad4

8 VolUp Messages Messages Dialpad5

9 VolDown DoNotDisturb DoNotDisturb Dialpad6

10 DialpadStar Hold MicMute VolDown

11 Dialpad0 VolUp VolUp n/a

12 DialpadPound VolDown VolDown Select

13 Dialpad9 DialpadPound DialpadPound Dialpad7

14 Dialpad8 Dialpad0 Dialpad0 Dialpad8

15 Dialpad7 DialpadStar DialpadStar Dialpad9

16 Dialpad4 Dialpad9 Dialpad9 MicMute

17 Dialpad5 Dialpad8 Dialpad8 n/a

18 Dialpad6 Dialpad7 Dialpad7 ArrowDown

19 Dialpad3 Dialpad4 Dialpad4 DialpadStar

20 Dialpad2 Dialpad5 Dialpad5 Dialpad0

21 Dialpad1 Dialpad6 Dialpad6 DialpadPound

22 n/a Dialpad3 Dialpad3 Redial

23 Redial Dialpad2 Dialpad2 n/a

24 n/a Dialpad1 Dialpad1 n/a

25 SoftKey3 SoftKey4 SoftKey4 Menu

26 MicMute SoftKey3 SoftKey3 Exit

27 SoftKey2 SoftKey2 SoftKey2 SoftKey1

28 SoftKey1 SoftKey1 SoftKey1 SoftKey2

29 ArrowDown Conference Services SoftKey3

30 n/a CallHistory Directories n/a

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Key ID IP 300 Function IP 500 Function IP 600 Function IP 4000 Function

31 ArrowUp Services Line6 n/a

32 Menu Directories Conference n/a

33 n/a Line3 Line2 n/a

34 n/a Line2 Line1 n/a

35 Headset Line1 Line3 n/a

36 n/a Redial Redial n/a

37 n/a Transfer Transfer n/a

38 n/a Headset Headset n/a

39 n/a MicMute Handsfree n/a

40 n/a Handsfree Hold n/a

41 n/a n/a Line4 n/a

42 n/a n/a Line5 n/a

3.1.8 Connected Party Identification

Where possible, the identity of the remote party to which the user has connected is displayed and logged . The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion.

3.1.9 Context Sensitive Volume Control

The volume of user interface sound effects, such as the ringer, and the receive volume of call audio is adjustable. While transmit levels are fixed according to the TIA/EIA-

810-A standard, receive volume is adjustable. For SoundPoint configuration parameters, the receive handset/headset volume resets to nominal after each call to comply with regulatory requirements. See 4.6.1.6.2 Volume Persistence <volume/> on page 80.

3.1.10 Customizable Audio Sound Effects

IP, if using the default

Audio sound effects used for incoming call alerting and other indications are customizable. Sound effects can be composed of patterns of synthesized tones or sample

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audio files. The default sample audio files may be replaced with alternates in .wav file format. Supported .wav formats include:

• mono G.711 (13-bit dynamic range, 8-khz sample rate),

• mono L16/1600 (16-bit dynamic range, 16-kHz sample rate).

Note

The alternate sampled audio sound effect files must be present on the boot server or the Internet for downloading at boot time.

Configuration File: ipmid.cfg

Specify patterns used for sound effects and the individual tones or sampled audio files used within them.

Central (boot server)

Web Server (if enabled)

Local

Local Phone User Interface

For more information, see:

• 4.6.1.1.3 Call Progress Tones <callProgTones> on page 68

• 4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> on page 72

• 4.6.1.5 Sound Effects <sound_effects/> on page 73

Specify sampled audio wave files to replace the built-in defaults. Navigate to:

http://<phoneIPAddress>/coreConf.htm#sa

Changes are saved to local flash and backed up to <Ethernet address>phone-.cfg on the boot server and will permanently override global settings unless deleted via the Reset User Set tings menu selection.

None.

3.1.11 Message Waiting Indication

The phone will flash a message-waiting indicator LED when instant messages are waiting, and it can be configured to do so when voice messages are waiting.

3.1.12 Distinctive Incoming Call Treatment

The phone can automatically apply distinctive treatment to calls containing specific attributes. The distinctive treatment that can be applied includes customizable alerting sound effects and automatic call diversion or rejection. Call attributes that can trigger distinctive treatment include the calling party name or SIP contact (number or URL format).

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Administration: Distinctive Incoming Call Treatment

For more information, see 3.1.16 Local Contact Directory on page 27.

3.1.13 Distinctive Ringing

There are three aspects to Distinctive Ringing:

1. The user can select the ring type for each line. There are many different ring patterns to choose from.

2. The ring type for specific callers can be assigned in the contact directory. For more information, see page 24. This feature has higher priority than Item 1.

3. The SIP Alert-Info field can be used to map calls to specific ring types. This feature has higher priority than Items 1 and 2.

3.1.12 Distinctive Incoming Call Treatment on

Central (boot server)

Local

Configuration file: sip.cfg

XML File: <Ethernet address>-directory.xml

Web Server (if enabled)

Local Phone User Interface

Specify the mapping of Alert-Info strings to ring types.

• For more information, see 4.6.2.1.3.2 Alert Infor-

This file can be created manually using an XML editor.

• For more information, see 3.1.16.1 Local Contact

None.

The user can edit the ring types selected for each line under the Settings menu. The user can also edit the directory contents.

Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. These changes will permanently override global settings unless deleted via the Reset User Settings menu selection.

3.1.13 Distinctive Call Waiting

mation <alertInfo/> on page 116.

Directory File Format on page 28.

The SIP Alert-Info field can be used to map calls to distinct call waiting types, currently limited to two styles.

Central (boot server)

Configuration file: sip.cfg

Specify the mapping of Alert-Info strings to call waiting types.

• For more information, see 4.6.2.1.3.2 Alert Information <alertInfo/> on page 116.

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Web S erv e r (if enabled)

Local

Local Phone User Interface

3.1.14 Do-Not-Disturb

A do-not-disturb feature is available to temporarily stop all incoming call alerting. Calls can optionally be treated as though the phone is busy while Do-Not-Disturb (DND) is enabled. Incoming calls received while DND is enabled are logged as missed.

Configuration file: ipmid.cfg

Central (boot server)

Configuration file: phone1.cfg

None.

Specify whether or not DND results in incoming calls being given busy treatment.

• For more information, see 4.6.1.10 Call Handling Configuration <call/> on page 92.

Specify whether DND is treated as a per-registration feature or a global feature on the phone.

• For more information, see 4.6.3.2.1 Do Not Disturb <donotdisturb/> on page 123.

Local

Web Server (if enabled)

Local Phone User Interface

None.

Enable or disable DND using the “Do Not Disturb” key on the IP 300, 500 and 600 or the Features menu on the IP 4000.

3.1.15 Handset, Headset, and Speakerphone

SoundPoint® IP phones come standard with a handset and a dedicated connector is provided for a headset (not supplied). The IP 500 and IP 600 phones have full-duplex

speakerphones. The IP 300 has a listen-only speakerphone. The SoundPoint provide dedicated keys for convenient selection of either the speakerphone or headset.

The SoundStation

Central (boot server)

IP 4000 phones are full-duplex speakerphones.

Configuration file: ipmid.cfg

Enable or disable persistent headset mode.

• For more information, see 4.6.1.2 User Preferences <user_preferences/> on page 69.

phones

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Local

Web Server (if enabled)

Local Phone User Interface

None.

Enable or disable persistent headset mode via the Settings menu. Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. Changes will permanently override global settings unless deleted via the Reset User Settings menu.

3.1.16 Local Contact Directory

The phone maintains a local contact directory. The directory can be downloaded from the boot server and edited locally. Contact information from previous calls may be easily added to the directory for convenient future access. The directory is the central database for several other features including speed-dial, distinctive incoming call treatment, presence, and instant messaging.

Configuration file: ipmid.cfg

Set whether the directory uses volatile storage on the phone (required on the IP 500 platform for directories greater than 25 entries).

Central (boot server)

XML file: 000000000000-directory.xml

XML file: <Ethernet address>-directory.xml

• For more information, see 4.6.1.11 Directory <directory/> on page 93.

A sample file named 000000000000-directory~.xml (Note extra “~” in the file name) is included with the application file distribution. This file can be used as a template for the per-phone <Ethernet address>-directory.xml directories (edit contents then rename to <Ethernet address>-directory.xml). It also can be used to seed new phones with an initial directory (edit contents than remove “~” from file name). Telephones without a local directory, such as new units from the factory, will download the 00000000000-directory.xml directory and base their initial directory on it. These files should be edited with an XML editor.

• For information on file format, see 3.1.16.1 Local Contact Directory File Format on page 28.

This file can be created manually using an XML editor.

• For information on file format, see 3.1.16.1 Local Contact Directory File Format on page 28.

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Local

Web Server (if enabled)

Local Phone User Interface

None.

The user can edit the directory contents at will. Changes will be stored in the phone’s flash file system and backed up to the boot server copy of <Ethernet address>-directory.xml if this is configured. When the phone boots, the boot server copy of the directory, if present, will overwrite the local copy.

3.1.16.1 Local Contact Directory File Format

An example local contact directory is shown. Look to the table for an explanation of each element.

Local Contact Directory File example:

<?xml version="1.0" encoding="UTF-8" standalone="yes" ?> <directory>

<item_list>

<item>

</item>

• • •

<item>

<ln>Smith</ln> <fn>Bill</fn> <ct>1003</ct> <sd>3</sd> <rt>3</rt> <dc /> <ad>0</ad> <ar>0</ar> <bw>0</bw> <bb>0</bb>

</item>

</item_list>

</directory>

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Element Permitted Values Interpretation

fn UTF-8 encoded string of up to

40 bytes

ln UTF-8 encoded string of up to

first name

last name

40 bytes

ct UTF-8 encoded string contain-

ing digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL

contact

Cannot be Null or duplicated; is used by the phone to address a remote party in the same way that a string of digits or a SIP URL are dialed manually by the user. This element is also used to associate incoming callers with a particular directory entry.

sd Null, 1 to 40 speed-dial index

Associates a particular entry with a speed dial bin for one-touch dialing or dialing from the speed dial menu.

rt Null, 1 to 21 ring type

When incoming calls can be associated with a directory entry by matching the address fields, this field is used to specify ring type to be used.

dc UTF-8 encoded string contain-

ing digits (the user part of a SIP URL) or a string that constitutes

divert contact

The forward-to address for the autodivert feature.

a valid SIP URL

ad 0,1 auto divert

If 1, automatically diverts callers that match the directory entry to the address specified in divertcontact.

ar 0,1

auto-reject

If 1, automatically rejects callers that match the directory entry.

bw 0,1 buddywatching

If 1, add this contact to the list of watched phones.

bb 0,1 buddyblock

If 1, block this contact from watching this phone.

a. In some cases, this will be less than 40 characters due to UTF-8’s variable length encoding. b. If auto-divert is also enabled, it has precedence over auto-reject.

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3.1.17 Local Digit Map

The phone has a local digit map feature to automate the setup phase of number-only calls. When properly configured, this feature eliminates the need for using the soft key when making outgoing calls. Instead, as soon as a digit pattern matching the digit map is found, the call setup process will complete automatically. This feature is similar to the digit map feature of the Media Gateway Control Protocol (MGCP) and the configuration syntax is the same as that specified in 2.1.5 of RFC 3435. The phone behavior when the user dials digits that do not match the digit map is configurable. It is also possible to strip a trailing # from the digits sent.

Send

Central (boot server)

Local

Configuration file: sip.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Specify impossible match behavior, trailing # behavior, digit map matching strings, and time out value.

• For more information, see 4.6.2.1.3.5 Conference Setup <conference/> on page 118.

Specify per-registration impossible match behavior, trailing # behavior, digit map matching strings, and time out values that override those in sip.cfg.

• For more information, see 4.6.3.4 Dial Plan <dialplan/> on page 126.

Specify digit map matching strings and time out value. Navigate to: http://<phoneIPAddress>/appConf.htm#ls

None.

3.1.18 Microphone Mute

A microphone mute feature is provided. When activated, visual feedback is provided. This is a local function and cannot be overridden by the network.

3.1.19 Multiple Call Appearances

The phone supports multiple concurrent calls. The hold feature can be used to pause activity on one call and switch to another call. When active on one call, an additional incoming call is presented using the familiar “call waiting” style. Soft keys with call disposition options are presented to the user. The current user interface is limited to

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two concurrent calls per registration (line). This is an artificial limit and will be expanded in the future.

3.1.20 Shared Call Appearances

Calls and lines on multiple phones can be logically related to each other. A call that is active on one phone will be presented visually to phones which share that call appearance. Mutual exclusion features emulate traditional PBX or key system privacy for shared calls. Incoming calls can be presented to multiple phones simultaneously. This feature is dependent on support from a SIP server which binds the appearances together logically and looks after the necessary state notifications and performs an access control function. For more information, see 5.2.4 Shared Call Appearance Signaling on page 139.

Central (boot server)

Configuration file: ipmid.cfg

Configuration file: sip.cfg

Configuration file: phone1.cfg

Specify whether diversion should be disabled on shared lines.

• For more information, see 4.6.1.10 Call Handling Configuration <call/> on page 92.

Specify line-seize subscription period.

• For more information, see 4.6.2.1.2 Server <server/> on page 114.

Specify standard or non-standard behavior for processing line-seize subscription for mutual exclusion feature.

• For more information, see 4.6.2.1.3.4 Special Events <specialEvent/> on page 118.

Specify per-registration line type (private or shared) and line-seize subscription period if using per-registration servers. A shared line will subscribe to a server providing call state information.

• For more information, see 4.6.3.1 Registration <reg/> on page 121.

Specify per-registration whether diversion should be disabled on shared lines.

• For more information, see 4.6.3.3 Diversion <divert/> on page 124.

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Local

Web S erve r (if enabled)

Local Phone User Interface

Specify line-seize subscription period. Navigate to:

http://<phoneIPAddress>/appConf.htm#se

Specify standard or non-standard behavior for processing line-seize subscription for mutual exclusion feature. Navigate to:

http://<phoneIPAddress>/appConf.htm#ls

Specify per-registration line type (private or shared) and line-seize subscription period if using per-registration servers, and whether diversion should be disabled on shared lines. Navigate to:

http://<phoneIPAddress>/reg.htm

Changes are saved to local flash and backed up to <Ether- net address>-phone.cfg on the boot server. They will permanently override global settings unless deleted via the Reset User Settings menu selection.

Specify per-registration line type (private or shared) using the SIP Configuration menu. Either the Web Server or the boot server configuration files or the local phone user interface should be used to configure registrations, not a mixture of these options. When the SIP Configuration menu is used, it is assumed that all registrations use the same server.

3.1.21 Bridged Line Appearances

Calls and lines on multiple phones can be logically related to each other. A call that is active on one phone will be presented visually to phones which share that line. Mutual exclusion features emulate traditional PBX or key system privacy for shared calls. Incoming calls can be presented to multiple phones simultaneously. This feature is dependent on support from a SIP server which binds the appearances together logically and looks after the necessary state notifications and performs an access control

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function. For more information, see 5.2.5 Bridged Line Appearance Signaling on page 139.

Note: In the configuration files, bridged lines are configured by “shared line” parameters.

Central (boot server)

Local

Configuration file: ipmid.cfg

Configuration file: phone1.cfg

Web S erve r (if enabled)

Specify whether diversion should be disabled on shared lines.

• For more information, see 4.6.1.10 Call Handling Configuration <call/> on page 92.

Specify per-registration line type (private or shared) and the shared line third party name. A shared line will subscribe to a server providing call state information.

• For more information, see 4.6.3.1 Registration <reg/> on page 121.

Specify per-registration whether diversion should be disabled on shared lines.

• For more information, see 4.6.3.3 Diversion <divert/> on page 124.

Specify per-registration line type (private or shared) and third party name, and whether diversion should be disabled on shared lines. Navigate to:

http://<phoneIPAddress>/reg.htm

3.1.22 Customizable Fonts and Indicators

The phone’s user interface can be customized by changing the fonts and graphic icons used on the display and the LED indicator patterns. Pre-existing fonts embedded in the software can be overwritten or new fonts can be downloaded. The bitmaps and bit-

Local Phone User Interface

Specify per-registration line type (private or shared) and the shared line third party name using the SIP Configuration menu. Either the Web Server or the boot server configuration files or the local phone user interface should be used to configure registrations, not a mixture of these options. When the SIP Configuration menu is used, it is assumed that all registrations use the same server.

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map animations used for graphic icons on the display can be changed and repositioned. LED flashing sequences and colors can be changed.

Configuration File: ipmid.cfg

Central (boot server)

Web S erv e r (if enabled)

Local

Local Phone User Interface

Specify fonts to overwrite existing ones or specify new fonts.

• For more information, see 4.6.1.12 Fonts <font/> on page 94.

Specify which bitmaps to use.

• For more information, see 4.6.1.14 Bitmaps <bitmaps/> on page 100.

Specify how to create animations and LED indicator patterns.

• For more information, see 4.6.1.15 Indicators <indicators/ > on page 100.

None.

3.1.23 Soft Key-Driven User Interface

The user interface makes extensive use of intuitive, context-sensitive soft key menus.

3.1.24 Speed Dial

Entries in the local directory can be linked to the speed dial system. The speed dial system allows calls to be placed quickly from dedicated keys as well as from a speed dial menu.

XML file: <Ethernet address>-directory.xml

Central (boot server)

The <sd>x</sd> element in the <Ethernet address>-directory.xml file links a directory

entry to a speed dial resource within the phone. Speed dial entries are mapped automatically to unused line keys ( line keys are not available on the IP 4000) and are available for selection within the speed dial menu. (Press the up-arrow key from the idle display to jump to SpeedDial).

• For more information, see 3.1.16.1 Local Contact Directory File Format on page 28.

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Web Server (if enabled) None.

Local Phone User Interface The user can edit the directory contents. The

Speed Dial Index field is used to link directory entries to speed dial operations.

Local

3.1.25 Time and Date Display

The phone maintains a local clock and calendar. Time and date can be displayed in certain operating modes such as when the phone is idle. The clock and calendar must be synchronized to a remote SNTP timeserver.

Configuration file: ipmid.cfg

Central (boot server)

Turn time and date display on or off.

• For more information, see 4.6.1.2 User Preferences

Set the time and date display formats.

• For more information, see 4.6.1.1.2 Date and Time

Set the basic SNTP settings.

Changes will be stored in the phone’s flash file system and backed up to the boot server copy of <Ethernet address>-directory.xml if this is configured. When the phone boots, the boot server copy of the directory, if present, will overwrite the local copy.

<user_preferences/> on page 69.

<datetime/> on page 68.

• For more information, see 4.6.1.8.2 Time Synchronization <SNTP/> on page 88.

Set daylight savings parameters.

• For more information, see 4.6.1.8.2 Time Synchronization <SNTP/> on page 88.

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Web S erv e r (if enabled)

Local Phone User Interface

Local

Set the basic SNTP and daylight savings settings.

Navigate to: http://<phoneIPAddress>/coreConf.htm#ti

Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. They will permanently override global settings unless deleted via the Reset User Settings menu selection.

The basic SNTP settings can be made in the Network Configuration menu.

• For more information, see 2.2.1.1 DHCP or Manual

The user can edit the time and date format under the Settings menu.

3.1.26 Idle Display Animation

TCP/IP Setup on page 5.

All phones except the SoundPoint® IP 300 can display a customized animation on the idle display in addition to the time and date. For example, a company logo could be displayed.

Configuration file: ipmid.cfg

Central (boot server)

Web S erv e r (if enabled)

Local

Local Phone User Interface

To turn idle display animation on or off.

• For more information, see 4.6.1.15 Indicators <indicators/> on page 100.

To replace the animation used for the idle display.

• For more information, see 4.6.1.15.1 Animations <Animations/> <IP_300/>, <IP_500/>, <IP_600/> and <IP_4000/> on page 101.

To change the position of the idle display animation.

• For more information, see 4.6.1.15.4.2 Graphic Icons <gi/> <IP_300/>, <IP_500/>, <IP_600/> and <IP_4000/> on page 102.

None.

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3.2 Call Management Features

3.2.1 Automatic Off-hook Call Placement

The phone supports an optional automatic off-hook call placement feature for each registration.

Central (boot server)

Local

3.2.2 Call Hold

Call hold is a fundamental feature of the phone. The purpose of hold is to pause activity on one call so that the user may use the phone for another task, such as to make or receive another call. Network signalling is employed to request that the remote party stop sending media and to inform them that they are being held. A configurable local hold reminder feature can be used to remind the user that they have placed calls on hold.

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Specify which registrations have the feature and what contact to call when going off hook.

• For more information, see 4.6.3.2.2 Automatic Offhook Call Placement <autoOffHook/> on page 123.

None.

Configuration file: sip.cfg

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Specify whether RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sendonly or a=inactive) outgoing hold signalling is used.

• For more information, see 4.6.2.1.3 SIP <SIP/> on page 115.

Specify local hold reminder options.

• For more information, see 4.6.1.10.1 Hold, Local Reminder <localReminder/> on page 93.

Specify whether or not to use RFC 2543 (c=0.0.0.0) outgoing hold signalling. The alternative is RFC 3264 (a=sendonly or a=inactive). .

Use the SIP Configuration menu to specify whether or not to use RFC 2543 (c=0.0.0.0) outgoing hold signalling. The alternative is RFC 3264 (a=sendonly or a=inactive).

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3.2.3 Call Transfer

Call transfer enables the user (User A or transferring user) to transform an existing call with User B (primary call) into a new call between User B and a third user C (transferred-to user) selected by User A. The phone offers both blind transfers (call is transferred to User C without first consulting privately with User C) and transfers with consultation (call is transferred to User C after first consulting privately with User C).

3.2.4 Three-Way Conference, Local or Centralized

Local or centralized conferences1 are supported. The phone can conference together the local user with the remote parties of two independent calls by using the phone’s local audio processing resources for the audio bridging. For a local conference there is no dependency on network signaling.

The phone also supports centralized conferences for which external resources are used such as a conference bridge. This depends on network signaling.

Central (boot server)

Local

Configuration file: sip.cfg

Web Server (if enabled)

Local Phone User Interface

Specify which type of conference to establish and the address of the centralized conference resource.

• For more information, see 4.6.2.1.3.5 Conference Setup <conference/> on page 118.

None.

3.2.5 Call Diversion (Call Forward)

The phone provides a flexible call diversion feature to divert (forward) calls to another destination. Call diversion can be applied automatically to all calls, only when the phone is busy, or after an extended period of alerting. The user can elect to manually divert calls while they are in the alerting state to a predefined or manually specified destination. The call diversion feature works in conjunction with the distinctive

1. On SoundStation IP® 4000, conferences are not available if the G.729 codec is enabled on the phone. This restriction will be removed in future releases.

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incoming call treatment feature. The user’s ability to originate calls is unaffected by all call diversion options. Each registration (line) has its own diversion properties.

Central (boot server)

Local

Configuration file: phone1.cfg

Web S erv e r (if enabled)

Local Phone User Interface

Set all call diversion settings including a global forward-to contact and individual settings for call forward all, call forward busy, call forward no-answer, and call forward do-notdisturb.

• For more information, see 4.6.3.3 Diversion <divert/> on page 124.

Set all call diversion settings.

Navigate to: http://<phoneIPAddress>/reg.htm

The user can set the call-forward-all setting from the idle display (enable/disable and specify the forward-to contact) as well as divert callers while the call is alerting.

3.2.6 Directed Call Pick-up

Calls to another phone can be picked up by dialing the extension of the other phone. This feature depends on support from a SIP server.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web S erve r (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, see 4.6.1.21 Feature <feature/> on page 109.

None.

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3.2.7 Group Call Pick-up

Calls to another phone within a pre-defined group can be picked up without dialing the extension of the other phone. This feature depends on support from a SIP server.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web S erve r (if enabled)

Local Phone User Interface

3.2.8 Call Park / Retrieve

An active call can be parked, and the parked call can be retrieved by another phone. This feature depends on support from a SIP server.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web S erve r (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, see 4.6.1.21 Feature <feature/> on page 109.

None.

Turn this feature on or off.

• For more information, see 4.6.1.21 Feature <feature/> on page 109.

None.

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3.2.9 Last Call Return

The phone allows server-based last call return. This feature depends on support from a SIP server.

Configuration file: ipmid.cfg

Central (boot server)

Web S erve r (if enabled)

Local

Local Phone User Interface

Turn this feature on or off.

• For more information, see 4.6.1.21 Feature <feature/> on page 109.

Specify the string sent to the server for last-call-return.

• For more information, see 4.6.1.10 Call Handling Configuration <call/> on page 92.

None.

3.3 Audio Processing Features

Proprietary state-of-the-art digital signal processing (DSP) technology is used to provide an excellent audio experience.

3.3.1 Low-Delay Audio Packet Transmission

The phone is designed to minimize latency for audio packet transmission.

3.3.2 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 (lost or excessively 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.

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Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Set the jitter buffer tuning parameters including minimum and maximum size and shrink aggression.

• For more information, see 4.6.1.6.1.2 Codec Profiles <profiles/> on page 79.

Set the jitter buffer tuning parameters including minimum and maximum size and shrink aggression.

Navigate to: http://<phoneIPAddress>/coreConf.htm#au

None.

3.3.3 Local Conference Mixing

The phone’s audio processing subsystem contains a flexible three-party conferencing

capability external protocol signaling is involved.

. This feature can be used to set up local three-party conferences where no

3.3.4 Voice Activity Detection (VAD)

The purpose of VAD is to conserve network bandwidth by detecting periods of relative “silence” in the transmit data path and replacing that silence efficiently with special packets that indicate silence is occurring. For those compression algorithms without an inherent VAD function, such as G.711, the phone is compatible with the comprehensive codec-independent comfort noise transmission algorithm specified in RFC 3389. This algorithm is derived from G.711 Appendix II, which defines a comfort noise (CN) payload format (or bit-stream) for G.711 use in packet-based, multimedia communication systems. The phone generates CN packets (also known as Silence Insertion Descriptor (SID) frames) and also decodes CN packets, efficiently regenerating a facsimile of the background noise at the remote end.

Central (boot server)

Configuration file: ipmid.cfg

Enable or disable VAD and set the detection threshold.

• For more information, see 4.6.1.6.10 Voice Activity Detection <VAD/> on page 86.

2. On SoundStation IP® 4000, conferences are not available if the G.729 codec is enabled on the phone. This restriction will be removed in future releases.

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Web Server (if enabled)

Local

Local Phone User Interface

None.

3.3.5 DTMF Tone Generation

The phone generates DTMF tones in response to user dialing on the dial pad. These tones are transmitted in the RTP streams of connected calls. The phone can encode the DTMF tones using the active voice codec or using RFC 2833 compatible encoding. The coding format decision is based on the capabilities of the remote endpoint.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Set the DTMF tone levels, autodialing on and off times, and other parameters.

• For more information, see 4.6.1.3.1 Dual Tone Multi-

None.

Frequency <DTMF/> on page 70.

3.3.6 DTMF Event RTP Payload

The phone is compatible with RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals. RFC 2833 describes a standard RTP-compatible tech-

nique for conveying DTMF dialing and other telephony events over an RTP media stream. The phone generates RFC 2833 (DTMF only) events but does not regenerate, nor otherwise use, DTMF events received from the remote end of the call.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Enable or disable RFC 2833 support in SDP offers and specify the payload value to use in SDP offers.

• For more information, see 4.6.1.3.1 Dual Tone MultiFrequency <DTMF/> on page 70.

None.

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3.3.7 Acoustic Echo Cancellation (AEC)

The phone employs advanced acoustic echo cancellation for hands-free operation. Both linear and non-linear techniques are employed to aggressively reduce echo yet provide for natural full-duplex communication patterns.

3.3.8 Audio Codecs

The following table summarizes the phone’s audio codec support:

Effective

Algorithm MIME Type Ref. Bit Rate

Sample Rate

Frame Size

audio bandwidth

G.711µ-law PMCU RFC

1890

G.711a-law PCMA RFC

1890

G.729AB G729 RFC

1890

SID CN RFC

3389

RFC 2833 phone-event RFC

2833

Configuration file: ipmid.cfg

Central (boot server)

Web S erv e r (if enabled)

64 Kbps 8 Ksps 10ms - 80ms 3.5KHz

8 Kbps 8 Ksps 10ms - 80ms 3.5KHz

N/A N/A N/A N/A

Specify codec priority, preferred payload sizes, and jitter buffer tuning parameters.

For more information, see

• 4.6.1.6.1.1 Codec Preferences <preferences/> on page 79, and

• 4.6.1.6.1.2 Codec Profiles <profiles/> on page 79.

Specify codec priority, preferred payload sizes, and jitter buffer tuning parameters.

Local

Local Phone User Interface

Navigate to: http://<phoneIPAddress>/coreConf.htm#au

None.

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3.3.9 Background Noise Suppression (BNS)

This feature, designed primarily for handsfree operation, reduces background noise to enhance communication in noisy environments.

3.3.10 Comfort Noise Fill

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 noise synthesis techniques to smooth out the noise level in the direction toward the remote user, providing a more natural call experience.

3.3.11 Automatic Gain Control (AGC)

This feature, applicable to handsfree operation, is used to boost the transmit gain of the

local talker in certain circumstances. and helps with the intelligibility of soft-talkers.

This increases the effective user-phone radius

3.4 Presence and Instant Messaging Features

The phone contains both Presence and Instant Messaging features. These features are compatible with Microsoft and Windows

Messenger 5.0. The phone’s presence and instant messaging features

are integrated with the contact directory features, using its contact database.

3.4.1 Presence

The Presence feature allows the phone to monitor the status of other users/devices and allows other users to monitor it. The status of monitored users is displayed visually and is updated in real time. The user can block others from monitoring her phone and

is notified when a change in monitored status occurs broadcast automatically to monitoring phones when the user engages in calls or

3. AGC support will be available in a subsequent release.

4. Notification when a change in monitored status occurs will be available in a subsequent release.

Windows® Messenger and MSN® Messenger version 4.7

. Phone status changes are

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invokes do-not-disturb. The user can also manually specify a state to convey, overriding, and perhaps masking, the automatic behavior.

XML file: <Ethernet

Central (boot server)

Local

address>-directory.xml

Web Server (if enabled)

Local Phone User Interface

3.4.2 Instant Messaging

The phone supports sending and receiving instant text messages. The user is alerted to incoming messages visually and audibly. The user can choose to view the messages immediately or when it is convenient. For sending messages, the user can choose to either select a message from a pre-set list of short messages, or an alphanumeric text entry mode allows the typing of custom messages using the dial pad. Message sending can be initiated by replying to an incoming message or by initiating a new dialog. The destination for new dialog messages can be entered manually or selected from the contact directory, the preferred method.

The <bw>0</bw> (buddy watching) and <bb>0</bb> (buddy blocking) elements in the <Ethernet address>directory.xml file dictate the Presence aspects of directory entries.

• For more information, see 3.1.16.1 Local Contact Directory File Format on page 28.

None.

The user can edit the directory contents. The Wat ch Buddy and Block Buddy fields control the buddy behavior of contacts.

3.5 Localization Features

3.5.1 Multilingual User Interface

All phones except SoundPoint® IP 300 have multilingual user interfaces. The System Administrator or the user can choose the language. Support for major western European languages is included and additional languages can be easily added. Support for

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Asian languages (Chinese, Japanese, and Korean) is also included but will render only

on the SoundPoint

IP 600’s and SoundStation® IP 4000’s higher resolution displays.

Basic character support includes the following Unicode character ranges:

Name Range

C0 Controls and Basic Latin U+0000 - U+007F

C1 Controls and Latin-1 Supplement U+0080 - U+00FF

Cyrillic (partial) U+0400 - U+045F

Extended character support available on SoundPoint® IP 600 and SoundStation® IP

4000 platforms includes the following Unicode character ranges

Name Range

CJK Symbols and Punctuation U+3000 - U+303F

Hiragana U+3040 - U+309F

Katakana U+30A0 - U+30FF

Bopomofo U+3100 - U+312F

Hangul Compatibility Jamo U+3130 - U+318F

Bopomofo Extended U+31A0 - U+31BF

Enclosed CJK Letters and Months U+3200 - U+327F

CJK Compatibility U+3300 - U+33FF

CJK Unified Ideographs U+4E00 - U+9FFF

Hangul Syllables U+AC00 - U+D7A3

CJK Compatibility Ideographs U+F900 - U+FAFF

CJK Half-width forms U+FF00 - U+FFFF

Note

The multilingual feature relies on dictionary files resident on the boot server. The dictionary files are downloaded from the boot server whenever the language is changed or at boot time when a language other than the internal US English language has been configured. If the dictionary files are inaccessible, the language will revert to the internal language.

5. Within a Unicode range, some characters may not be supported due to their infrequent usage.

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Note

Currently, the multilingual feature is only available in the application. At this time, the bootROM application is English only.

Central (boot server)

Configuration file: ipmid.cfg

Specify the boot-up language and the selection of language choices to be made available to the user.

For more information, see:

• 4.6.1.1.1 Multilingual <multilingual/> on page 66, and

• 4.6.1.1.1.1 Adding New Languages on page 67.

Local

Web Server (if enabled)

Local Phone User Interface

None.

The user can select the preferred language under the Settings menu. Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. Changes will permanently override global settings unless deleted via the Reset User Settings menu selection.

3.5.2 Downloadable Fonts

New fonts can be loaded onto the phone. For more information, see 4.6.1.12 Fonts <font/> on page 94.

3.5.3 Synthesized Call Progress Tones

In order to emulate the familiar and efficient audible call progress feedback generated by the PSTN and traditional PBX equipment, call progress tones are synthesized dur-

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ing the life cycle of a call. These call progress tones are easily configurable for compatibility with worldwide telephony standards or local preferences.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Specify the basic tone frequencies, levels, and basic repetitive cadences.

• For more information, see 4.6.1.3.2 Chord Sets <chord_sets/> on page 71 and 4.6.1.1.3 Call Progress Tones <callProgTones> on page 68.

Specify downloaded sampled audio files for advanced call progress tones.

• For more information, see 4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> on page 72.

Specify patterns.

For more information, see:

• 4.6.1.5.1 Patterns <patterns/> on page 74, and

• 4.6.1.5.1.1 Call Progress Patterns on page 75.

None.

3.6 Advanced Server Features

3.6.1 Voicemail Integration

The phone is compatible with voicemail servers. The subscribe contact and callback mode can be configured per user/registration on the phone. The phone can be configured with a SIP URL to be called automatically by the phone when the user elects to retrieve messages. Voicemail access can be configured to be via a single key press if only one registration has voicemail set up and the phone has a dedicated function key for this purpose (for example the Messages key on the IP 500 and IP 600). A message-

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waiting signal from a voicemail server will trigger the message-waiting indicator to flash.

Central (boot server)

Configuration file: ipmid.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

For one-touch voicemail access, enable the “one-touch voicemail” user preference.

• For more information, see 4.6.1.2 User Preferences <user_preferences/> on page 69.

For one-touch voicemail access, choose to bypass instant messages to remove the step of selecting between instant messages and voicemail after pressing the Messages key on the IP 500 and IP 600 (instant messages are still accessible from the Main Menu).

On a per-registration basis, specify a subscribe contact for solicited NOTIFY applications, a callback mode (self callback or another contact), and the contact to call when the user accesses voicemail.

• For more information, see 4.6.3.5 Messaging <msg/> on page 128.

For one-touch voicemail access, enable the “one-touch voicemail” user preference and choose to bypass instant messages to remove the step of selecting between instant messages and voicemail after pressing the Messages key on the IP 500 and IP 600 (instant messages are still accessible from the Main Menu).

Navigate to: http://<phoneIPAddress>/coreConf.htm

On a per-registration basis, specify a subscribe contact for solicited NOTIFY applications, a callback mode (self call-

Local

Local Phone User Interface

back or another contact) to call when the user accesses voicemail.

Navigate to: http://<phoneIPAddress>/reg.htm

None.

3.6.2 Multiple Registrations

SoundPoint® IP phones support multiple registrations per phone and the SoundSta-

tion

IP 4000 supports a single registration. The SoundPoint® IP 300 supports two registrations, the IP 500 supports three and the IP 600 supports six. Each registration is mapped to the familiar concept of a phone line. The user can select which line to

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use for outgoing calls or which registration to use when initiating new instant message dialogs.

Central (boot server)

Configuration file: sip.cfg

Configuration file: phone1.cfg

Specify the local SIP signaling port and an array of SIP servers to register to. For each server specify the registration period and the signaling failure behavior.

• For more information, see 4.6.2.1.1 Local <local/> on page 113 and 4.6.2.1.2 Server <server/> on page 114.

For up to six registrations, specify a display name, a SIP address, an optional display label, an authentication user ID and password, and an optional array of registration servers. The authentication user ID and password are optional and for security reasons can be omitted from the configuration files. The local flash parameters will be used instead. The optional array of servers and their associated parameters will override the servers specified in sip.cfg if non-Null.

• For more information, see 4.6.3.1 Registration <reg/> on page 121.

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Local

Web Server (if enabled)

Local Phone User Interface

Specify the local SIP signaling port and an array of SIP servers to register to.

Navigate to: http://<phoneIPAddress>/appConf.htm

For up to six registrations, specify a display name, a SIP address, an optional display label, an authentication user ID and password, and an optional array of registration servers. The authentication user ID and password are optional and for security reasons can be omitted from the configuration files. The local flash parameters will be used instead. The optional array of servers will override the servers specified in sip.cfg in non-Null. This will also override the servers on the appConf.htm web page.

Navigate to: http://<phoneIPAddress>/reg.htm

Use the SIP Configuration menu to specify the local SIP signaling port, an array of SIP servers to register to and registration information for up to six registrations. The SIP Configuration menu contains a sub-set of all the parameters available in the configuration files. Either the Web Server or the boot server configuration files or the local phone user interface should be used to configure registrations, not a mixture of these options. When the SIP Configuration menu is used, it is assumed that all registrations use the same server. Changes are saved to local flash and backed up to <Ethernet address>-phone.cfg on the boot server. Changes will permanently override global settings unless deleted via the Reset User Settings menu selection.

• For more information on the fields in this menu, see

4.6.2.1.1 Local <local/> on page 113, 4.6.2.1.2 Server <server/> on page 114 and 4.6.3.1 Registration <reg/> on page 121.

3.6.3 ACD login / logout

The phone allows ACD (Automatic Call Distribution) login and logout. This feature depends on support from a SIP server.

Central (boot server)

Configuration file: ipmid.cfg

Turn this feature on or off.

• For more information, see 4.6.1.21 Feature <feature/> on page 109.

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Web S erve r (if enabled)

Local

Local Phone User Interface

None.

3.6.4 ACD agent available / unavailable

The phone supports ACD (Automatic Call Distribution) agent available and unavailable. This feature depends on support from a SIP server.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web S erve r (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, see 4.6.1.21 Feature <feature/> on page 109.

None.

3.6.5 Server Redundancy

The phone can be configured with multiple SIP servers, one primary and one or more backup. The phone will switch to a backup server when the current primary server fails. Backup server configuration can be static or can use advanced DNS methods. In the case of static server lists, when a server registration fails, registration will be attempted on another server. If the phone is not registered to the first server in the list when registration fails, it will start by trying to register to the first server. When making a new call, if the INVITE fails, the other servers in the list will be tried one by one for routing signaling until the last server is tried.

Definition of signaling failure (registration or start of call): If TCP is used: The signaling fails if the connection fails or the Send fails. If UDP is used: The signaling fails if ICMP is detected or if the signal times out. If the signaling has been attempted via 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 4.6.2.1.2 Server <server/> on page 114 and

4.6.3.1 Registration <reg/> on page 121.

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3.6.5.1 DNS SIP Server Name Resolution

If a DNS name is given for a proxy/registrar address, the IP address(es) associated with that name will be discovered as specified in RFC 3263 - Locating SIP Servers. If a port is given, the only lookup will be an A record. If no port is given, NAPTR and SRV records will be 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, 5060 will be used.

See http://www.ietf.org/rfc/rfc3263.txt for an example.

3.7 Accessory Internet Features

3.7.1 MicroBrowser

The SoundPoint® IP 600 phone supports an XHTML microbrowser. This can be launched by pressing the Services key.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Specify the Services browser home page, a proxy to use, and size limits.

• For more information, see 4.6.1.23 MicroBrowser <microbrowser/> on page 111.

None

3.8 Security Features

3.8.1 Local User and Administrator Privilege Levels

Several local settings menus are protected with two privilege levels, user and administrator, each with its own password. The phone will prompt for either the user or administrator password before granting access to the various menu options. When the

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user password is requested, the administrator password will also work. The web server is protected by the administrator password.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Specify the minimum lengths for the user and administrator passwords.

• For more information, see 4.6.1.17.1 Password Lengths <pwd/><length> on page 106.

None.

The user and administrator passwords can be changed under the Settings menu. Passwords can consist of ASCII characters 32-127 (0x20-0x7F) only.

Changes are saved to local flash but are not backed up to <Ethernet address>-phone.cfg on the boot server for security reasons.

3.8.2 Incoming Signaling Validation

Three optional levels of security are provided for validating incoming network signaling:

• source IP address validation

• digest authentication

• both

Central (boot server)

Local

Configuration File: sip.cfg

Web Server (if enabled)

Local Phone User Interface

Specify the type of validation to perform on a request-byrequest basis, appropriate to specific event types in some cases.

• For more information, see 4.6.2.1.3.3 Request Validation <requestValidation/> on page 117.

None.

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4 Optimization

4.1 Ethernet Switch

The SoundPoint® IP phones contain two Ethernet ports, labeled LAN and PC, and an embedded Ethernet switch that runs at full line-rate. The Ethernet switch allows a personal computer and other Ethernet devices to connect to the office LAN by daisy chaining through the phone, eliminating the need for a stand-alone hub. The

SoundPoint phone. SoundPoint

powered (power supplied via the signaling or idle pairs of the LAN Ethernet cable). Line powering typically requires that the phone plugs directly into a dedicated LAN

jack. Devices that do not require LAN power can then plug into the SoundPoint PC Ethernet port.

IP switch gives higher transmit priority to packets originating in the

IP can be powered via a local AC power adapter or can be line-

SoundPoint® IP Switch - Port Priorities

To help ensure good voice quality, the Ethernet switch embedded in the

SoundPoint

IP phones should be configured to give voice traffic emanating from the phone higher transmit priority than those from a device connected to the PC port. If not using a VLAN (VLAN blank in the setup menu), this will automatically be the case. If using a VLAN, ensure that the 802.1p priorities for both default and RTP packet types are set to 2 or greater. Otherwise, these packets will compete equally with those from the PC port. For more information, see 4.6.1.7 Quality of Service <QOS/> on page 86.

4.2 Application Network Setup

4.2.1 RTP Ports

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 will negotiate the respective destination IP addresses and ports. This allows RTCP to operate correctly even with RTP media flowing in only a single direction, or not at all. It also allows greater security: packets from unauthorized sources can be rejected.

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The phone can filter incoming RTP packets arriving on a particular port by IP address. Packets arriving from a non-negotiated IP address can be discarded.

The phone can also enforce symmetric port operation for RTP packets: packets arriving with the source port set to other than the negotiated remote sink port can be rejected.

The phone can also jam the destination transport port to a specified value regardless of the negotiated port. This can be useful for punching through firewalls. When this is enabled, all RTP traffic will be sent to the specified port and will be expected to arrive on that port as well. Incoming packets are sorted by the source IP address and port, allowing multiple RTP streams to be multiplexed.

The RTP port range used by the phone can be specified. Since conferencing and multiple RTP streams are supported, several ports can be used concurrently. Consistent with RFC 1889, the next higher odd port is used to send and receive RTCP.

Configuration file: ipmid.cfg

Central (boot server)

Web S erv e r (if enabled)

Local

Local Phone User Interface

Specify whether to filter incoming RTP packets by IP address, whether to require symmetric port usage, whether to jam the destination port and specify the local RTP port range start.

• For more information, see 4.6.1.8.3.1 RTP <RTP/> on page 90.

Specify whether to filter incoming RTP packets by IP address, whether to require symmetric port usage, whether to jam the destination port and specify the local RTP port range start.

Navigate to: http://<phoneIPAddress>/coreConf.htm#rt

None.

4.2.2 Working with Network Address Translation (NAT)

The phone can work with certain types of network address translation (NAT). The phone’s signaling and RTP traffic use symmetric ports (the source port in transmitted packets is the same as the associated listening port used to receive packets) and the external IP address and ports used by the NAT on the phone’s behalf can be configured on a per-phone basis.

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Central (boot server)

Local

Configuration file: phone1.cfg

Web S erv e r (if enabled)

Local Phone User Interface

Specify the external NAT IP address and the ports to be used for signaling and RTP traffic.

• For more information, see 4.6.3.6 Network Address Translation <nat/> on page 129.

Specify the external NAT IP address and the ports to be used for signaling and the RTP traffic.

Navigate to: http://<phoneIPAddress>/coreConf.htm#na

None.

4.3 Updating and Rebooting

The bootROM, application executable, and configuration files can be updated automatically via the centralized provisioning (boot server) model.

To automatically update:

1. Back-up old application and configuration files. The old configuration can be easily restored by reverting to the back-up files.

2. Customize new configuration files or apply new or changed parameters to the old configuration files. Differences between old and new versions of configuration files are explained in the Release the software.Changes to site-wide configuration files such as ipmid.cfg can be done manually, but a scripting tool is useful to change per-phone configuration files.

3. Save the new configuration files and images (such as sip.ld) on the boot

server.

4. Reboot the phones (see below).

6. For TFTP, refer to 2.2.2.1.1 FTP vs. TFTP on page 10

Notes which accompany

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For more information, see 2.2.2 Application Configuration on page 9.

Manual Reboot: Menu Option or Key Presses

To reboot phones manually, a menu option can be selected or a key combination can be held down. The menu option is called Restart Phone and it is in the Settings menu. For the key combination, press and hold the following keys simultaneously until a confirmation tone is heard or for about three seconds:

IP 300: Volume-, Volume+, Hold, Do Not Disturb

IP 500: Volume-, Volume+, Hold, Messages

IP 600: Volume-, Volume+, Mute, Messages

IP 4000: *, #, Volume+, Select

Centralized Reboot:

The phones can be rebooted remotely via the SIP signaling protocol. Refer to

4.6.2.1.3.4 Special Events <specialEvent/> on page 118

Periodic Polling For Upgrades:

The phones can be configured to periodically poll the boot server to check for changed configuration files or application executable. If a change is detected the phone will reboot to download the change. Refer to

page 107.

4.4 Event Logging

The phones maintain both boot and application event log files. These files can be helpful when diagnosing problems. The event log files are stored in the phone’s flash file system and are periodically uploaded to the provisioning boot server if permitted by security policy. The files are stored in the phone’s home directory or a user-configurable directory on the boot server. Both overwrite and append modes are supported for the application event log.

The event log files are:

• <Ethernet address>-boot.log

• <Ethernet address>-app.log

The boot log file is uploaded to the boot server after every reboot.

4.6.1.18 Provisioning <provisioning/> on

The application log file is uploaded periodically or when the local copy reaches a predetermined size.

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As an additional diagnostic tool, both log files can be uploaded on demand to the boot server by pressing and holding the following keys until a confirmation tone is heard or for about three seconds.

IP 300: Line1, Line2, Arrow Up, Arrow Down

IP 500: The four arrow keys

IP 600: The four arrow keys

IP 4000: Menu, Exit, Off-hook/Handsfree, Redial

Log files uploaded in this manner are named:

• <Ethernet address>-now-boot.log

• <Ethernet address>-now-app.log

Central (boot server)

Local

Configuration file: ipmid.cfg

Configuration file: sip.cfg

Configuration file: <Ethernet address>.cfg

Web Server (if enabled)

Local Phone User Interface

Specify a multitude of event logging settings.

• For more information, see 4.6.1.16 Event Logging <logging/> on page 103.

Specify “sip” event logging settings.

• For more information, see 4.6.2.1.5 SIP Logging <logging/> <level/> <change/> on page 120.

Specify different directory to use for log files if desired.

• For more information, see 2.2.2.1.2.1 Master Configuration Files on page 10.

Specify a multitude of event logging settings.

Navigate to: http://<phoneIPAddress>/coreConf.htm#lo

Specify “sip” event logging settings.

Navigate to: http://<phoneIPAddress>/appConf.htm#lo

None.

4.5 Audio Quality Issues and VLANs

The phone contains both network layer and Ethernet layer support for prioritizing voice and signaling traffic over the network. Quality of Service (QoS) parameters include IP type-of-service (TOS) bits, and Ethernet IEEE 802.1p user priority. These can be set on a per-protocol basis. The phone also supports RTCP per RFC 1889.

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4.5.1 IP TOS

The “type of service” field in an IP packet header consists of four TOS bits and a 3-bit precedence field. Each TOS bit can be set to either 0 or 1. The precedence field can be set to a value from 0 through 7. The type of service can be configured specifically for RTP packets and call control packets, such as SIP signaling packets.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

4.5.2 IEEE 802.1p/Q

The phone will tag all Ethernet packets it transmits with an 802.1Q VLAN header when it has a valid VLAN ID set in its network configuration, or is instructed to tag packets via Cisco Discovery Protocol (CDP) running on a connected Ethernet switch, or a VLAN ID is obtained from DHCP (see 2.2.1.2.2 DHCP Menu on page 7). The

802.1p/Q user_priority field can be set to a value from 0 to 7. The user_priority can be configured specifically for RTP packets and call control packets, such as SIP signaling packets, with default settings configurable for all other packets.

Specify protocol-specific IP TOS settings.

• For more information, see 4.6.1.7.2 IP TOS <IP/> on page 87.

Specify IP TOS settings.

Navigate to: http://<phoneIPAddress>/coreConf.htm#qo

None.

Central (boot server)

Local

Configuration file: ipmid.cfg

Web Server (if enabled)

Local Phone User Interface

Specify default and protocol-specific 802.1p/Q settings.

• For more information, see 4.6.1.7.1 Ethernet IEEE

802.1p/Q <Ethernet/> on page 86.

Specify 802.1p/Q settings.

Navigate to http://<phoneIPAddress>/coreConf.htm#qo

Specify whether CDP is to be used or manually set the VLAN ID or configure DHCP VLAN Discovery.

Phase 1: bootRom - Navigate to: SETUP menu during autoboot countdown.

Phase 2: Application - Navigate to: Menu>Settings>Network Configuration

• For more information, see 2.2.1 Basic Network Setup on page 4.

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4.5.3 RTCP Support

The phone supports RTCP per RFC 1889. For each RTP stream, which, by convention, uses even ports only, the next higher odd port is used to send and receive RTCP reports.

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4.6 Configuration Files

This section is a reference for all parameters that are configurable when using the centralized provisioning installation model. It is divided into three sections:

• Core Configuration - ipmid.cfg

• Application Configuration - sip.cfg

• Per-phone Configuration - phone1.cfg

Notes

In the following tables, “Null” should be interpreted as the empty string, that is, attributeName=“” when the file is viewed in a text editor. To enter special characters in a configuration file, enter the appropriate sequence using a text editor. See the table below.

Special Character Required Character Sequence in Text Editor

” "

’ '

4.6.1 Core Configuration - ipmid.cfg

4.6.1.1 Localization <localization/>

The phone has a multilingual user interface. It supports both North American and international time and date formats. The call progress tones can also be customized. For more information, see 4.6.1.1.3 Call Progress Tones <callProgTones> on page 68,

4.6.1.3.2 Chord Sets <chord_sets/> on page 71, and 4.6.1.5.1.1 Call Progress Patterns on page 75.

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4.6.1.1.1 Multilingual <multilingual/>

The multilingual feature is based on string dictionary files downloaded from the boot server. These files are encoded in standalone XML format. Several western European and Asian languages are included with the distribution.

Permitted

Attribute

Values

Interpretation

lcl.ml.lang Null

An exact match for one of the folder names under the SoundPointIPLocalization folder on the boot server.

lcl.ml.lang.menu.x String in the format

language_region

lcl.ml.lang.cpt.x positive integer The call progress tone index to be

If Null, the default internal language (US English) will be used, otherwise, the language to be used may be specified in the format language-region.

Multiple lcl.ml.lang.menu.x attributes are supported - as many languages as are desired. However, the lcl.ml.lang.menu.x attributes must be sequential (lcl.ml.lang.menu.1, lcl.ml.lang.menu.2, lcl.ml.lang.menu.3, ..., lcl.ml.lang.menu.N) with no gaps and the strings must exactly match a folder name under the SoundPointIPLocalization folder on the boot server for the phone to be able to locate the dictionary file.

associated with this language. See

4.6.1.1.3 Call Progress Tones <callProgTones> on page 68.

lcl.ml.lang.clock.x.24HourClock 0,1 If attribute present, overrides

lcl.datetime.time.24HourClock;

If 1, display time in 24-hour clock mode rather than a.m./p.m.

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Permitted

Attribute

Values

Interpretation

lcl.ml.lang.clock.x.format string which

includes ‘D’, ‘d’ and ‘M’ and two optional comma’s

lcl.ml.lang.clock.x.longFormat 0, 1 If attribute present, overrides

lcl.ml.lang.clock.x.dateTop 0, 1 If attribute present, overrides

lcl.ml.lang.y.list “All” or a comma-

separated list

If attribute present, overrides lcl.datetime.date.format; D = day of week d = day M = month Up to two comma’s may be included. e.g. D,dM = Thursday, 3 July Md,D = July 3, Thursday The field may contain 0, 1 or 2 comma’s which can occur only between characters and only one at a time i.e. “D,,dM” is illegal.

lcl.datetime.date.longFormat;

If 1, display the day and month in long format (Friday/November), otherwise use abbreviations (Fri/Nov).

lcl.datetime.date.dateTop;

If 1, display date above time, otherwise display time above date.

A list of the languages supported on hardware platform ‘y’ where ‘y’ can be IP_500 or IP_600.

4.6.1.1.1.1 Adding New Languages

Follow these steps to add new languages to those included with the distribution:

1. Create a new dictionary file based on an existing one.

2. Change the strings making sure to encode the XML file in UTF-8 but also ensuring the UTF-8 characters chosen are within the Unicode character ranges indicated in

3. Place the file in an appropriately named folder according to the format language_region parallel to the other dictionary files under the SoundPoint IPLocalization folder on the boot server.

4. Add a lcl.ml.lang.clock.menu.x attribute to the configuration file.

5. Add lcl.ml.lang.clock.x.24HourClock, lcl.ml.lang.clock.x.format, lcl.ml.lang.clock.x.longFormat and lcl.ml.lang.clock.x.dateTop attributes and set them according to the regional preferences.

6. (Optional) Set lcl.ml.lang to be the new language_region string.

3.5.1 Multilingual User Interface on page 46.

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4.6.1.1.2 Date and Time <datetime/>

Permitted

Attribute

lcl.datetime.time.24HourClock 0,1 If 1, display time in 24-hour clock mode rather

Values

Interpretation

than a.m./p.m.

lcl.datetime.date.format string which

includes ‘D’, ‘d’ and ‘M’ and two optional comma’s

lcl.datetime.date.longFormat 0,1 If 1, display the day and month in long format

lcl.datetime.date.dateTop 0, 1 If 1, display date above time else display time

Controls format of date string. D = day of week d = day M = month Up to two comma’s may be included. e.g. D,dM = Thursday, 3 July Md,D = July 3, Thursday The field may contain 0, 1 or 2 comma’s which can occur only between characters and only one at a time i.e. “D,,dM” is illegal.

(Friday/November), otherwise, use abbreviations (Fri/Nov).

above date.

4.6.1.1.3 Call Progress Tones <callProgTones>

Call progress tone overrides can be used to customize the tones for a particular country or region. The overrides set offered by default spans all default languages on the phone. Tone overrides are based on the ITU-T Recommendation E.180 Supplement 2 entitled Telephone Network and ISDN - Operation, numbering, routing and mobile service - Various tones used in national networks.

Attribute

lcl.cpt positive

lcl.cpt.menu.x string String to specify the country or region e.g. Italy.

Permitted Values

integer OR blank

Interpretation

The index of the default tone overrides to be selected by the phone. If blank, default call progress tones are used.

Multiple lcl.cpt.menu.x strings are supported, the strings are displayed in the Call Progress Tones menu. The lcl.cpt.menu.x attributes must be sequential (lcl.cpt.menu.1, lcl.cpt.menu.2, lcl.cpt.menu.3, ..., lcl.cpt.menu.N) with no gaps.

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In the following table, x is the index of the region as specified by the x index of the lcl.cpt.menu.x attribute above, y is the chord set number and cat is one of cp or misc. For more information, see 4.6.1.5.1.1 Call Progress Patterns on page 75.

Permitted

Attribute

lcl.cpt.chord.cat.x.y.freq.z 0-1600 Frequency for this component in Hertz; up to four

lcl.cpt.chord.cat.x.y.level.z -57 to 3 Level of this component in dBm0.

Values Interpretation

chord-set components can be specified (z=1, 2, 3,

4).

lcl.cpt.chord.cat.x.y.onDur positive

integer

lcl.cpt.chord.cat.x.y.offDur positive

integer

lcl.cpt.chord.cat.x.y.repeat positive

integer

On duration in milliseconds, 0=infinite.

Off duration in milliseconds, 0=infinite.

Specifies how many times the ON/OFF cadence is repeated, 0=infinite.

4.6.1.2 User Preferences <user_preferences/>

Permitted

Attribute

up.headsetMode 0,1 0 If set to 1, the headset will be selected as

up.useDirectoryNames 0,1 0 If set to 1, the name fields of directory

Values

Default Interpretation

the preferred transducer after its first use until the headset key is pressed again; otherwise, handsfree will be selected preferentially over the headset.

entries which match incoming calls will be used for caller identification display and in the call lists instead of the name provided via network signaling.

up.oneTouchVoiceMail 0, 1 0 If set to 1, the voicemail summary dis-

up.welcomeSoundEnabled 0, 1 1 If set to 1, play welcome sound effect

up.welcomeSoundOnWarmBootEnabled

play is bypassed and voicemail is dialed directly (if configured).

after a reboot.

0, 1 0 If set to 1, play welcome sound effect on

warm as well as cold boots, otherwise only a cold boot will trigger the welcome sound effect.

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Permitted

Attribute

up.localClockEnabled 0, 1 1 If set to 1, display the date and time on

Values

Default Interpretation

the idle display

4.6.1.3 Tones <tones/>

This section describes configuration items for the tone resources available in the phone.

4.6.1.3.1 Dual Tone Multi-Frequency <DTMF/>

Permitted

Attribute

Values

Default Interpretation

tone.dtmf.level -33 to -3 -15 Level of the high frequency compo-

nent of the DTMF digit measured in dBm0; the low frequency tone will be two dB lower.

tone.dtmf.onTime positive

integer

tone.dtmf.offTime positive

integer

tone.dtmf.chassis.masking 0, 1 0 If set to 1, DTMF tones will be sub-

50 When a sequence of DTMF tones is

played out automatically, this is the length of time in milliseconds the tones will be generated for; this is also the minimum time the tone will be played for when dialing manually (even if key press is shorter).

50 When a sequence of DTMF tones is

played out automatically, this is the length of time in milliseconds the phone will pause between digits; this is also the minimum inter-digit time when dialing manually.

stituted with a non-DTMF pacifier tone when dialing in hands-free mode. This prevents DTMF digits being broadcast to other surrounding telephony devices or being inadvertently transmitted in-band due to local acoustic echo.

Note: tone.dtmf.chassis.masking should only be enabled when tone.dtmf.viaRtp is disabled.

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Permitted

Attribute

tone.dtmf.stim.pac.offHookOnly 0, 1 0 Not currently used.

tone.dtmf.viaRtp 0, 1 1 If set to 1, encode DTMF in the

tone.dtmf.rfc2833Control 0, 1 1 If set to 1, the phone will indicate a

Values

Default Interpretation

active RTP stream, otherwise, DTMF may be encoded within the signaling protocol only when the protocol offers the option.

Note: tone.dtmf.chassis.masking should be enabled when tone.dtmf.viaRtp is disabled.

preference for encoding DTMF via RFC 2833 format in its Session Description Protocol (SDP) offers by showing support for the phoneevent payload type; this does not affect SDP answers, these will always honor the DTMF format present in the offer since the phone has native support for RFC 2833.

tone.dtmf.rfc2833Payload 96-127 101 The phone-event payload encoding

4.6.1.3.2 Chord Sets <chord_sets/>

Chord sets are the building blocks of sound effects that use synthesized rather than sampled audio (most call progress and ringer sound effects). A chord-set is a multifrequency note with an optional on/off cadence. A chord-set can contain up to four frequency components generated simultaneously, each with its own level.

There are three blocks of chord sets:

• callProg: used for call progress sound effect patterns

• ringer

• misc (miscellaneous)

All three blocks use the same chord set specification format.

in the dynamic range to be used in SDP offers.

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In the following table, x is the chord-set number and cat is one of callProg, ringer, or misc.

Permitted

Attribute

tone.chord.cat.x.freq.y 0-1600 Frequency for this component in Hertz; up to four

tone.chord.cat.x.level.y -57 to 3 Level of this component in dBm0.

Values Interpretation

chord-set components can be specified (y=1, 2, 3,

4).

tone.chord.cat.x.onDur positive

integer

tone.chord.cat.x.offDur positive

integer

tone.chord.cat.x.repeat positive

integer

On duration in milliseconds, 0=infinite.

Off duration in milliseconds, 0=infinite.

Specifies how many times the ON/OFF cadence is repeated, 0=infinite.

4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/>

The following sampled audio WAVE file (.wav) formats are supported:

• mono 8 kHz G.711 µ-Law

• G.711 A-Law

• L16/1600 (16-bit, 16 kHz sampling rate, mono)

The phone uses built-in wave files for some sound effects. The built-in wave files can be replaced with files downloaded from the boot server or from the Internet, however, these are stored in volatile memory so the files will need to remain accessible should the phone need to be rebooted. Files will be truncated to a maximum size of 300 kilobytes.

In the following table, x is the sampled audio file number.

Attribute Permitted Values Interpretation

saf.x Null OR valid path

name OR an RFC 1738-compliant URL to a HTTP, FTP, or TFTP wave file resource.

Note: Refer to the above wave file format restrictions.

If Null, the phone will use a built-in file;

If set to a path name, the phone will attempt to download this file at boot time from the boot server;

If set to a URL, the phone will attempt to download this file at boot time from the Internet.

Note: A TFTP URL is expected to be in the format: tftp://<host>/[pathname]<filename>, for example: tftp:// somehost.example.com/sounds/example.wav

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The following table defines the default usage of the sampled audio files with the phone:

Sampled Audio File Default use within phone (pattern reference)

1 Welcome Sound Effect (se.pat.misc.7)

2 Ringer 13 (se.pat.ringer.13)

3 Ringer 14 (se.pat.ringer.14)

4 Ringer 15 (se.pat.ringer.15)

5 Ringer 16 (se.pat.ringer.16)

6 Ringer 17 (se.pat.ringer.17)

7 Ringer 18 (se.pat.ringer.18)

8 Ringer 19 (se.pat.ringer.19)

9 Ringer 20 (se.pat.ringer.20)

10 Ringer 21 (se.pat.ringer.21)

11 Ringer 22 (se.pat.ringer.22)

12-24 Not used.

4.6.1.5 Sound Effects <sound_effects/>

The phone uses both synthesized (based on the chord-sets described earlier) and sampled audio sound effects. Sound effects are defined by patterns: rudimentary sequences of chord-sets, silence periods, and wave files.

Permitted

Attribute

se.stutterOnVoiceMail 0, 1 1 If set to 1, stuttered dial tone is used in place

se.appLocalEnabled 0, 1 1 If set to 1, local user interface sound effects

Values

Default Interpretation

of normal dial tone to indicate that one or more messages (voice-mail) are waiting at the message center.

such as confirmation/error tones, will be enabled.

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4.6.1.5.1 Patterns <patterns/>

Patterns use a simple script language that allows different chord sets or wave files to be strung together with periods of silence. The script language uses the following instructions:

Instruction Meaning Example

sampled (n) Play sampled audio

file n

chord (n, d) Play chord set n (d is

optional and allows the chord set ON duration to be overridden to d milliseconds)

silence (d) Play silence for d

milliseconds (Rx audio is not muted)

branch (n) Advance n instruc-

tions and execute that instruction (n must be negative and must not branch beyond the first instruction)

se.pat.callProg.x.inst.y.type =”sampled” (sampled audio file instruction type)

se.pat.callProg.x.inst.y.value =”3” (specifies sampled audio file 3)

se.pat.callProg.x.inst.y.type = “chord” (chord set instruction type)

se.pat.callProg.x.inst.y.value = “3” (specifies call progress chord set 3)

se.pat.callProg.x.inst.y.param = “2000” (override ON duration of chord set to 2000 milliseconds)

se.pat.callProg.x.inst.y.type = “silence” (silence instruction type)

se.pat.callProg.x.inst.y.value = “300” (specifies silence is to last 300 milliseconds)

se.pat.callProg.x.inst.y.type = “branch” (branch instruction type)

se.pat.callProg.x.inst.y.value = “-5” (step back 5 instructions and execute that instruction)

a. Currently, patterns that use the sampled instruction are limited to the following format:

sampled followed by optional silence and optional branch back to the beginning.

In the following table, x is the pattern number, y is the instruction number. Both x and y need to be sequential. There are three categories of sound effect patterns: callProg (call progress patterns), ringer and misc (miscellaneous).

Attribute Permitted Values Interpretation

se.pat.callProg.x.name UTF-8 encoded

string

Used for identification purposes in the user interface (currently used for ringer patterns only); for patterns that use a sampled audio file which has been overridden by a downloaded replacement, the se.pat.ringer.x.name parameter will be overridden in the user interface by the file names of the wave file.

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Attribute Permitted Values Interpretation

se.pat.callProg.x.inst.y.type sampled OR chord

se.pat.callProg.x.inst.y.value integer Instruction type:

se.pat.callProg.x.inst.y.param positive integer If instruction type is chord, this optional parameter

4.6.1.5.1.1 Call Progress Patterns

The following table maps call progress patterns to their usage within the phone.

Call progress pattern number

Use within phone

OR silence OR branch

As above.

Interpretation:

sampled

chord

silence

branch

specifies the on duration to be used, overriding the on duration specified in the chord-set definition.

sampled audio file number

chord set number

silence duration in ms

number of instructions to advance

1 dial tone

2 busy tone

3 ring back tone

4 reorder tone

5 stuttered dial tone

6 call waiting tone

7 alternate call waiting tone (distinctive)

8 confirmation tone

9 howler tone (off-hook warning)

10 record warning

11 message waiting tone

12 alerting

13 intercom announcement tone

14 barge-in tone

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4.6.1.5.1.2 Ringer Patterns

The following table maps ringer pattern numbers to their default descriptions.

Ringer pattern number

Default description

Silent Ring

2 Low Trill

3 Low Double Trill

4 Medium Trill

5 Medium Double Trill

6 High Trill

7 High Double Trill

8 Highest Trill

9 Highest Double Trill

10 Beeble

11 Triplet

12 Ringback-style

Sampled audio file 2

14 Sampled audio file 3

15 Sampled audio file 4

16 Sampled audio file 5

17 Sampled audio file 6

18 Sampled audio file 7

19 Sampled audio file 8

20 Sampled audio file 9

21 Sampled audio file 10

22 Sampled audio file 11

a. Silent Ring will only provide a visual indication of an incoming call, but no audio

indication.

b. Sampled audio files 1-21 all use the same built-in file unless that file has been

replaced with a downloaded file. For more information, see 4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> on page 72.

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4.6.1.5.1.3 Miscellaneous Patterns

The following table maps miscellaneous patterns to their usage within the phone.

Miscellaneous pattern number

1 new message waiting indication

2 new instant message

3 not used

4 local hold notification

5 positive confirmation

6 negative confirmation

7 welcome (boot up)

Use within phone

4.6.1.5.2 Ring type <ringType/>

Ring type is used to define a simple class of ring to be applied based on some credentials that are usually carried within the network protocol. The ring class includes attributes such as call-waiting and ringer index, if appropriate. The ring class can use one of four types of ring that are defined as follows:

ring Play a specified ring pattern or call waiting indication

visual Provide only a visual indication (no audio indication) of incoming call (no

ringer needs to be specified).

answer

ring-answer

a. Note that auto-answer on incoming call is currently only applied if there is no other

call in progress on the phone at the time.

b. See note a.

In the following table, x is the ring class number. The x index needs to be sequential.

Provide auto-answer on incoming call

Provide auto answer on incoming call after a ring period

Attribute Permitted Values Interpretation

se.rt.enabled 0,1 Set to 1 to enable the ring type feature within

the phone, 0 otherwise.

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Attribute Permitted Values Interpretation

se.rt.modification.enabled 0,1 Set to 1 to allow user modification via local

user interface of the pre-defined ring type

enabled for modification

se.rt.x.name UTF-8 encoded string Used for identification purposes in the user

interface

se.rt.x.type ring OR visual OR

answer OR ringanswer

se.rt.x.ringer integer - only relevant

if the type is set to ‘ring’ or ‘ring-answer’

se.rt.x.callWait integer - only relevant

if the type is set to ‘ring’ or ‘ring-answer’

se.rt.x.timeout positive integer - only

relevant if the type is set to ‘ring-answer’. Default value is 2000.

se.rt.x.mod 0,1 Set to 1 if the user interface should allow for

a. Modification via user interface will be implemented in a future release. b. Modification via user interface will be implemented in a future release.

As defined in table above.

The ringer index to be used for this class of ring. The ringer index should match one of

4.6.1.5.1.2 Ringer Patterns on page 76.

The call waiting index to be used for this class of ring. The call waiting index should match one defined in Progress Patterns on page 75.

The duration of the ring in milliseconds before the call is auto answered. If this field is omitted or is left blank, a value of 2000 is used.

modification by the user of the ringer index used for this ring class.

4.6.1.5.1.1 Call

4.6.1.6 Voice Settings <voice/>

4.6.1.6.1 Voice Coding Algorithms <codecs/>

The following voice codecs are supported:

MIME

Algorithm

G.711µ-law PMCU G711mu 64 Kbps 8 Ksps 10ms - 80ms 3.5KHz

G.711a-law PCMA G711A 64 Kbps 8 Ksps 10ms - 80ms 3.5KHz

G.729AB G729 G729AB 8 Kbps 8 Ksps 10ms - 80ms 3.5KHz

Type

Label Bit Rate

Sample Rate

Frame Size

Effective Audio Bandwidth

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4.6.1.6.1.1 Codec Preferences <preferences/>

Permitted

Attribute

voice.codecPref.G711Mu Null, 1-3 1 Specifies the codec preferences for

voice.codecPref.G711A 2

voice.codecPref.G729AB 3

voice.codecPref.IP_4000.G711Mu Null, 1-3 1 Specifies the codec preferences for

voice.codecPref.IP_4000.G711A 2

Values

Default Interpretation

SoundPoint

1=highest 3=lowest Null=do not use

Give each codec a unique priority, this will dictate the order used in SDP negotiations.

Note

specified, the phone will automatically use G711Mu and G711A with default parameters.

the SoundStation Interpretation as above.

: If less than two codecs are

IP platforms.

IP 4000 platform.

voice.codecPref.IP_4000.G729AB Null Not supported by default so that

4.6.1.6.1.2 Codec Profiles <profiles/>

The following profile attributes can be adjusted for each of the three supported codecs. In the table, x=G711Mu, G711A, or G729AB.

Attribute

voice.audioProfile.x.payloadSize 10, 20, 30,

Permitted Values

...80

G.711Mu and G.711A local conferences can be supported. This restriction will be removed in a future release.

Interpretation

Preferred Tx payload size in milliseconds to be provided in SDP offers and used in the absence of ptime negotiations. This is also the range of supported Rx payload sizes.

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Permitted

Attribute

Values

Interpretation

voice.audioProfile.x.jitterBufferMin 40, 50, 60, ...

(multiple of

10)

voice.audioProfile.x.jitterBufferShrink 10, 20, 30, ...

(multiple of

10)

voice.audiProfile.x.jitterBufferMax > jitterBuf-

ferMin, multiple of 10, <=500 for IP 500 and IP 600, <= 160 for IP 300

The smallest jitter buffer depth (in milliseconds) that must be achieved before play out begins for the first time. Once this depth has been achieved initially, the depth may fall below this point and play out will still continue. This parameter should be set to the smallest possible value which is at least two packet payloads, and larger than the expected short term average jitter.

The absolute minimum duration time (in milliseconds) of RTP packet Rx with no packet loss between jitter buffer size shrinks. Use smaller values (1000 ms) to minimize the delay on known good networks. Use larger values to minimize packet loss on networks with large jitter (3000 ms).

The largest jitter buffer depth to be supported (in milliseconds). Jitter above this size will always cause lost packets. This parameter should be set to the smallest possible value that will support the expected network jitter.

4.6.1.6.2 Volume Persistence <volume/>

The user’s selection of the receive volume during a call can be remembered between calls. This can be configured per termination (handset, headset and handsfree/chassis). In some countries regulations exist which dictate that receive volume should be reset to nominal at the start of each call on handset and headset.

Permitted

Attribute

voice.volume.persist.handset 0, 1 0 If set to 1, the receive volume will be

voice.volume.persist.headset 0, 1 0

voice.volume.persist.handsfree 0, 1 1

Values

Default Interpretation

remembered between calls. If set to 0, the receive volume will be reset to nominal at the start of each call.

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4.6.1.6.3 Gains <gains/>

The default gain settings have been carefully adjusted to comply with the TIA-810-A digital telephony standard.

Do not alter these values.

Attribute Default

voice.gain.rx.analog.handset 0

voice.gain.rx.analog.headset 0

voice.gain.rx.analog.chassis 3

voice.gain.rx.analog.chassis.IP300 -6

voice.gain.rx.analog.chassis.IP_4000 3

voice.gain.rx.analog.ringer 3

voice.gain.rx.analog.ringer.IP300 -6

voice.gain.rx.analog.ringer.IP_4000 3

voice.gain.rx.digital.handset -15

voice.gain.rx.digital.headset -21

voice.gain.rx.digital.chassis 0

voice.gain.rx.digital.chassis.IP_4000 0

voice.gain.rx.digital.ringer -21

voice.gain.rx.digital.ringer.IP_4000 -21

voice.gain.rx.analog.handset.sidetone -24

voice.gain.rx.analog.headset.sidetone -24

voice.gain.tx.analog.handset 3

voice.gain.tx.analog.headset 3

voice.gain.tx.analog.chassis 3

voice.gain.tx.analog.chassis.IP_4000 3

voice.gain.tx.digital.handset 0

voice.gain.tx.digital.headset 0

voice.gain.tx.digital.chassis 6

voice.gain.tx.digital.chassis.IP_4000 0

voice.gain.tx.analog.preamp.handset 23

voice.gain.tx.analog.preamp.headset 23

voice.gain.tx.analog.preamp.chassis 42

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4.6.1.6.4 Acoustic Echo Cancellation <AEC/>

These settings control the performance of the speakerphone acoustic echo canceller.

Do not alter these values.

Attribute Default

voice.aec.hs.enable 0

voice.aec.hs.lowFreqCutOff 125

voice.aec.hs.highFreqCutOff 3800

voice.aec.hs.erlTab_0_300 -24

voice.aec.hs.erlTab_300_600 -24

voice.aec.hs.erlTab_600_1500 -24

voice.aec.hs.erlTab_1500_3500 -24

voice.aec.hs.erlTab_3500_7000 -24

voice.aec.hd.enable 0

voice.aec.hd.lowFreqCutOff 125

voice.aec.hd.highFreqCutOff 3800

voice.aec.hd.erlTab_0_300 -24

voice.aec.hd.erlTab_300_600 -24

voice.aec.hd.erlTab_600_1500 -24

voice.aec.hd.erlTab_1500_3500 -24

voice.aec.hd.erlTab_3500_7000 -24

voice.aec.hf.enable 1

voice.aec.hf.lowFreqCutOff 125

voice.aec.hf.highFreqCutOff 3800

voice.aec.hf.erlTab_0_300 -6

voice.aec.hf.erlTab_300_600 0

voice.aec.hf.erlTab_600_1500 0

voice.aec.hf.erlTab_1500_3500 -6

voice.aec.hf.erlTab_3500_7000 -6

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4.6.1.6.5 Acoustic Echo Suppression <AES/>

These settings control the performance of the speakerphone acoustic echo suppressor.

Do not alter these values.

Attribute Default

voice.aes.hs.enable 0

voice.aes.hs.duplexBalance 7

voice.aes.hd.enable 0

voice.aes.hd.duplexBalance 0

voice.aes.hf.enable 1

voice.aes.hf.duplexBalance.0 7

voice.aes.hf.duplexBalance.1 7

voice.aes.hf.duplexBalance.2 6

voice.aes.hf.duplexBalance.3 6

voice.aes.hf.duplexBalance.4 6

voice.aes.hf.duplexBalance.5 4

voice.aes.hf.duplexBalance.6 4

voice.aes.hf.duplexBalance.7 3

voice.aes.hf.duplexBalance.8 2

voice.aes.hf.duplexBalance.IP_4000.0 10

voice.aes.hf.duplexBalance.IP_4000.1 9

voice.aes.hf.duplexBalance.IP_4000.2 8

voice.aes.hf.duplexBalance.IP_4000.3 7

voice.aes.hf.duplexBalance.IP_4000.4 6

voice.aes.hf.duplexBalance.IP_4000.5 5

voice.aes.hf.duplexBalance.IP_4000.6 4

voice.aes.hf.duplexBalance.IP_4000.7 3

voice.aes.hf.duplexBalance.IP_4000.8 2

4.6.1.6.6 Background Noise Suppression <NS/>

These settings control the performance of the transmit background noise suppression feature.

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Do not alter these values.

Attribute Default

voice.ns.hs.enable 0

voice.ns.hs.signalAttn -6

voice.ns.hs.silenceAttn -8

voice.ns.hd.enable 0

voice.ns.hd.signalAttn 0

voice.ns.hd.silenceAttn 0

voice.ns.hf.enable 1

voice.ns.hf.signalAttn -6

voice.ns.hf.silenceAttn -8

voice.ns.hf.IP_4000.enable 1

voice.ns.hf.IP_4000.signalAttn -6

voice.ns.hf.IP_4000.silenceAttn -8

4.6.1.6.7 Automatic Gain Control <AGC/>

These settings control the performance of the transmit automatic gain control feature.

Do not alter these values.

Attribute Default

voice.agc.hs.enable 0

voice.agc.hd.enable 0

voice.agc.hf.enable 0

7. Automatic Gain Control will be implemented in a future release.

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4.6.1.6.8 Receive Equalization <RXEQ/>

These settings control the performance of the handsfree receive equalizaton feature.

Do not alter these values.

Attribute Default

voice.rxEq.hf.IP_500.enable 1

voice.rxEq.hf.IP_600.enable 1

voice.rxEq.hf.IP_4000.enable 0

voice.rxEq.hf.type 0

voice.rxEq.hf.IP_4000.type 0

voice.rxEq.hf.size 63

voice.rxEq.hf.IP_4000.size 33

voice.rxEq.hf.coef.0 to .62 Don’t change these values

voice.rxEq.hf.coef.IP_4000.0 to .32 Don’t change these values

4.6.1.6.9 Transmit Equalization <TXEQ/>

These settings control the performance of the handsfree transmit equalization feature.

Do not alter these values.

Attribute Default

voice.txEq.hf.IP_500.enable 0

voice.txEq.hf.IP_600.enable 0

voice.txEq.hf.IP_4000.enable 0

voice.txEq.hf.coef.1 to 20 Don’t change these values

voice.txEq.hf.coef.IP_4000.1 to 20 Don’t change these values

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4.6.1.6.10 Voice Activity Detection <VAD/>

These settings control the performance of the voice activity detection (silence suppression) feature.

Permitted

Attribute

voice.vadEnable 0, 1 0 If set to 1, enable VAD.

Values

Default Interpretation

voice.vadThresh integer from

0 to 30

15 The threshold for determining what is active

voice and what is background noise in dB. This does not apply to G.729AB codec operation which has its own built-in VAD function.

4.6.1.7 Quality of Service <QOS/>

These settings control the Quality of Service (QOS) options.

4.6.1.7.1 Ethernet IEEE 802.1p/Q <Ethernet/>

These settings control the 802.1p/Q user_priority field.

4.6.1.7.1.1 RTP <RTP/>

These parameters apply to RTP packets.

Permitted

Attribute

Values

Default Interpretation

qos.ethernet.rtp.user_priority 0-7 5 User-priority used for RTP packets.

4.6.1.7.1.2 Call Control <CallControl/>

These parameters apply to call control packets, such as the network protocol signaling.

Attribute

qos.ethernet.callControl.user_priority 0-7 5 User-priority used for call con-

Permitted Values

Default Interpretation

trol packets.

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4.6.1.7.1.3 Other <Other/>

These default parameter values are used for all packets which are not set explicitly.

Permitted

Attribute

qos.ethernet.other.user_priority 0-7 2 User-priority used for packets that

Values

Default Interpretation

do not have a per-protocol setting.

4.6.1.7.2 IP TOS <IP/>

These settings control the “type of service” field in outgoing packets.

4.6.1.7.2.1 RTP <RTP/>

These parameters apply to RTP packets.

Permitted

Attribute

qos.ip.rtp.min_delay 0, 1 1 If set to 1, set min-delay bit in

qos.ip.rtp.max_throughput 0, 1 1 If set to 1, set max-throughput

qos.ip.rtp.max_reliability 0, 1 0 If set to 1, set max-reliability

qos.ip.rtp.min_cost 0, 1 0 If set to 1, set min-cost bit in

qos.ip.rtp.precedence 0-7 5 If set to 1, set precedence bits

Values

Default Interpretation

the IP TOS field of the IP header, or else don’t set it.

bit in the IP TOS field of the IP header, or else don’t set it.

the IP TOS field of the IP header, or else don’t set it.

in the IP TOS field of the IP header, or else don’t set them.

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4.6.1.7.2.2 Call Control <CallControl/>

These parameters apply to call control packets, such as the network protocol signaling.

Permitted

Attribute

qos.ip.callControl.min_delay 0, 1 1 If set to 1, set min-delay bit in

qos.ip.callControl.max_throughput 0, 1 0 If set to 1, set max-throughput

qos.ip.callControl.max_reliability 0, 1 0 If set to 1, set max-reliability bit

qos.ip.callControl.min_cost 0, 1 0 If set to 1, set min-cost bit in the

Values

Default Interpretation

the IP TOS field of the IP header, or else don’t set it.

bit in the IP TOS field of the IP header, or else don’t set it.

in the IP TOS field of the IP header, or else don’t set it.

IP TOS field of the IP header, or else don’t set it.

qos.ip.callControl.precedence 0-7 5 If set to 1, set precedence bits in

4.6.1.8 Basic TCP/IP <TCP_IP/>

4.6.1.8.1 Network Monitoring <netMon/>

Do not alter these values.

Attribute Permitted Values Default

tcpIpApp.netMon.enabled 0, 1 1

tcpIpApp.netMon.period 1 to 86400 30

4.6.1.8.2 Time Synchronization <SNTP/>

The following table describes the parameters used to set up time synchronization and daylight savings time. The defaults shown will enable daylight savings time for North America.

the IP TOS field of the IP header, or else don’t set them.

Daylight savings defaults:

• don’t use fixed day, use first or last day of week in the month,

• start DST on the first Sunday in April at 2 am,

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• stop DST on the last Sunday in October at 2 am.

Permitted

Attribute

Values

Default Interpretation

tcpIpApp.sntp.resyncPeriod positive

integer

tcpIpApp.sntp.address

valid host name or IP

86400 (24 hours)

clock Address of the SNTP

Time in seconds between SNTP resyncs.

server.

address

tcpIpApp.sntp.gmtOffset positive or

negative integer

-28800 (Pacific time)

Offset in seconds of the local time zone from GMT.

Note: 3600 seconds per hour

tcpIpApp.sntp.daylightSavings.enable 0, 1 1 If set to 1, apply day-

light savings rules to displayed time.

tcpIpApp.sntp.daylightSavings.fixedDayEnable

0, 1 0 If set to 1, “April 1st” is

used, otherwise “the first Sunday in April” is used.

tcpIpApp.sntp.daylightSavings.start.month 1-12 4 (April) Month to start DST.

1=Jan, 2=Feb, ..., 12=Dec

tcpIpApp.sntp.daylightSavings.start.date 1-31 1 Day of the month to

start DST.

tcpIpApp.sntp.daylightSavings.start.time 0-23 2 Time of day to start

DST, in 24 hour clock. 2=2 am, 14=2 pm

tcpIpApp.sntp.daylightSavings.start.dayOfWeek

1-7 1 Day of week to apply

DST. 1=Sun, 2=Mon, ..., 7=Sat

tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth

0, 10 0 If set to 1 and fixedDay-

Enable=0, start DST on the last day of the week (specified by dayOfWeek) in the month, rather than the first in the month.

tcpIpApp.sntp.daylightSavings.stop.month 1-12 10 Month to stop DST.

1=Jan, 2=Feb, ..., 12=Dec

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Permitted

Attribute

tcpIpApp.sntp.daylightSavings.stop.date 1-31 1 Day of the month to

tcpIpApp.sntp.daylightSavings.stop.time 0-23 2 Time of day to stop

Values

Default Interpretation

start DST.

DST, in 24 hour clock. 2= 2 am, 14=2 pm

tcpIpApp.sntp.daylightSavings.stop.dayOfWeek

tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth

a. Both tcpIpApp.sntp.address and tcpIpApp.sntp.gmtOffset can be provided via DHCP. If so,

the DHCP parameters will override the parameters in ipmid.cfg.

4.6.1.8.3 port <port/>

4.6.1.8.3.1 RTP <RTP/>

Attribute

1-7 1 Day of week to stop

DST. 1=Sun, 2=Mon, ..., 7=Sat

0, 1 1 If set to 1 and fixedDay-

Enable=0, stop DST on the last day of the week (specified by dayOfWeek) in the month, rather than the first in the month.

Permitted Values

Default Interpretation

tcpIpApp.port.rtp.filterByIp 0, 1 1 If set to 1, reject RTP pack-

tcpIpApp.port.rtp.filterByPort 0, 1 0 If set to 1, reject RTP pack-

tcpIpApp.port.rtp.forceSend Null, 1024-

65534

ets arriving from (sent from) a non-negotiated (via SDP) IP address.

ets arriving from (sent from) a non-negotiated (via SDP) port.

Null When non-Null, send all

RTP packets to, and expect all RTP packets to arrive on, the specified port. Note: both tcpIpApp.port.rtp.filterByIp and tcpIpApp.port.rtp.filterByPort must be enabled for this to work.

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Permitted

Attribute

Values

Default Interpretation

tcpIpApp.port.rtp.mediaPortRangeStart Null, even

integer from 1024-65534

4.6.1.9 Web Server <HTTPD/>

The phone contains a local web server for user and administrator features. This can be disabled for applications where it is not needed or where it poses a security threat. The web server supports both basic and digest authentication. The authentication user name and password are not configurable for this release.

Permitted

Attribute

Values

Default Interpretation

Null If set to Null, the value 2222

will be used for the first allocated RTP port, otherwise, the specified port will be used. Subsequent ports will be allocated from a pool starting with the specified port plus two up to a value of (start-port + 46), after which the port number will wrap back to the starting value.

httpd.enabled 0, 1 1 If set to 1, the HTTP server will be enabled.

4.6.1.9.1 Configuration <cfg/>

Permitted

Attribute

httpd.cfg.enabled 0, 1 1 If set to 1, the HTTP server configuration

httpd.cfg.port 1-65535 80 Port is 80 for HTTP servers. Care should be

Values

Default Interpretation

interface will be enabled.

taken when choosing an alternate port.

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4.6.1.10 Call Handling Configuration <call/>

Permitted

Attribute

call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with

call.enableOnNotRegistered 0, 1 1 If set to 1, calls will be allowed when the

Values

Default Interpretation

the reason “busy” if do-not-disturb is enabled.

phone is not successfully registered, otherwise, calls will not be permitted without a valid registration.

call.offeringTimeOut positive

integer

call.ringBackTimeOut positive

integer

call.lastCallReturnString string of

maximum length 32

call.callWaiting.prompt 0, 1 0 If set to 1, an incoming call received when

call.shared.disableDivert 0, 1 1 If set to 1, disable diversion feature for

call.shared.seizeFailReorder 0, 1 1 If set to 1, play re-order tone locally on

call.shared.oneTouchResume

0, 1 0 If set to 1, when a shared line has a call on

60 Time in seconds to allow an incoming call

to ring before dropping the call, 0=infi-

nite

60 Time in seconds to allow an outgoing call

to remain in the ringback state before dropping the call, 0=infinite.

*69 The string sent to the server when the user

selects the “last call return” action.

another call is active will change the User Interface focus (call appearance and soft keys).

shared lines.

shared line seize failure.

hold the remote user can press that line and resume the call. If more than one call is on hold on the line then the first one will be selected and resumed automatically. If set to 0, pressing the shared line will bring up a list of the calls on that line and the user can select which call the next action should be applied to.

a. The call diversion, no answer feature will take precedence over this feature if enabled. For

more information, see 4.6.3.3.3 No Answer <noanswer/> on page 125.

Polycom Version 1.4.x 17, SoundPoint IP 300, SoundPoint IP 500, SoundPoint IP 600, SoundStation IP 4000 Administrator's Manual

Specifications and Main Features

Frequently Asked Questions

User Manual