Polycom SOUNDPOINT IP 300, SOUNDPOINT IP 501, SOUNDPOINT IP 301 User Manual

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Administrator Guide

SoundPoint®/SoundStation® IP SIP

Version 1.6.x

18 July 2005

Notices

1. Specifications subject to change without notice.

Polycom, Inc.

1565 Barber Lane, Milpitas CA 95035, USA

www.polycom.com

Part Number: 1725-11530-160 Rev A

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 Provisioning File Transfer ......................................................................................................5

2.2.1.3 Local User Interface Setup Menus ..........................................................................................7

2.2.1.4 Reset to Factory Defaults......................................................................................................11

2.2.2 Application Configuration................................................................................................................ 11

2.2.2.1 Centralized Configuration......................................................................................................11

2.2.2.2 Local Phone Configuration.....................................................................................................17

3 Features .......................................................... 19

3.1 Basic Features.........................................................19

3.1.1 Call Log ......................................................................................................................................... 19

3.1.2 Call Timer ...................................................................................................................................... 19

3.1.3 Call Waiting ................................................................................................................................... 20

3.1.4 Called Party Identification............................................................................................................... 20

3.1.5 Calling Party Identification.............................................................................................................. 20

3.1.6 Missed Call Notification................................................................................................................... 20

3.1.7 Configurable Feature Keys.............................................................................................................. 21

3.1.8 Connected Party Identification ........................................................................................................ 25

3.1.9 Context Sensitive Volume Control.................................................................................................... 25

3.1.10 Customizable Audio Sound Effects................................................................................................. 25

3.1.11 Message Waiting Indication .......................................................................................................... 26

3.1.12 Distinctive Incoming Call Treatment............................................................................................... 26

3.1.13 Distinctive Ringing........................................................................................................................ 27

3.1.14 Distinctive Call Waiting ................................................................................................................. 27

3.1.15 Do-Not-Disturb............................................................................................................................. 28

3.1.16 Handset, Headset, and Speakerphone........................................................................................... 28

3.1.17 Local Contact Directory................................................................................................................. 29

3.1.17.1 Local Contact Directory File Format.......................................................................................31

3.1.18 Local Digit Map............................................................................................................................ 32

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

3.1.19 Microphone Mute ......................................................................................................................... 33

3.1.20 Multiple Line Keys per Registration ............................................................................................... 33

3.1.21 Multiple Call Appearances............................................................................................................. 34

3.1.22 Shared Call Appearances .............................................................................................................. 35

3.1.23 Bridged Line Appearances............................................................................................................. 37

3.1.24 Customizable Fonts and Indicators................................................................................................. 38

3.1.25 Soft Key-Driven User Interface...................................................................................................... 39

3.1.26 Speed Dial ................................................................................................................................... 39

3.1.27 Time and Date Display.................................................................................................................. 40

3.1.28 Idle Display Animation ................................................................................................................. 41

3.2 Call Management Features........................................42

3.2.1 Automatic Off-hook Call Placement................................................................................................. 42

3.2.2 Call Hold........................................................................................................................................ 43

3.2.3 Call Transfer .................................................................................................................................. 43

3.2.4 Three-Way Conference, Local or Centralized.................................................................................... 44

3.2.5 Call Diversion (Call Forward) .......................................................................................................... 44

3.2.6 Directed Call Pick-up ...................................................................................................................... 45

3.2.7 Group Call Pick-up.......................................................................................................................... 46

3.2.8 Call Park / Retrieve ....................................................................................................................... 46

3.2.9 Last Call Return.............................................................................................................................. 47

3.3 Audio Processing Features ........................................47

3.3.1 Low-Delay Audio Packet Transmission............................................................................................. 47

3.3.2 Jitter Buffer and Packet Error Concealment ..................................................................................... 47

3.3.3 Local Conference Mixing................................................................................................................. 48

3.3.4 Voice Activity Detection (VAD)......................................................................................................... 48

3.3.5 DTMF Tone Generation ................................................................................................................... 49

3.3.6 DTMF Event RTP Payload ................................................................................................................ 49

3.3.7 Acoustic Echo Cancellation (AEC) ..................................................................................................... 50

3.3.8 Audio Codecs.................................................................................................................................. 50

3.3.9 Background Noise Suppression (BNS).............................................................................................. 51

3.3.10 Comfort Noise Fill......................................................................................................................... 51

3.3.11 Automatic Gain Control (AGC)....................................................................................................... 51

3.4 Presence and Instant Messaging Features ...................51

3.4.1 Presence........................................................................................................................................ 51

3.4.2 Instant Messaging .......................................................................................................................... 52

3.5 Localization Features ...............................................52

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

3.5.1 Multilingual User Interface ............................................................................................................. 52

3.5.2 Downloadable Fonts ....................................................................................................................... 54

3.5.3 Synthesized Call Progress Tones...................................................................................................... 54

3.6 Advanced Server Features ........................................55

3.6.1 Voicemail Integration ..................................................................................................................... 55

3.6.2 Multiple Registrations..................................................................................................................... 56

3.6.3 ACD login / logout ......................................................................................................................... 59

3.6.4 ACD agent available / unavailable.................................................................................................. 59

3.6.5 Server Redundancy ........................................................................................................................ 59

3.6.5.1 DNS SIP Server Name Resolution...........................................................................................60

3.7 Accessory Internet Features.......................................60

3.7.1 MicroBrowser................................................................................................................................. 60

3.8 Security Features.....................................................61

3.8.1 Local User and Administrator Privilege Levels.................................................................................. 61

3.8.2 Custom Certificates......................................................................................................................... 61

3.8.3 Incoming Signaling Validation......................................................................................................... 62

4 Optimization .................................................... 63

4.1 Ethernet Switch .......................................................63

4.2 Application Network Setup .......................................63

4.2.1 RTP Ports....................................................................................................................................... 63

4.2.2 Working with Network Address Translation (NAT) ............................................................................64

4.3 Updating and Rebooting...........................................65

4.4 Event Logging .........................................................66

4.5 Audio Quality Issues and VLANs ................................67

4.5.1 IP TOS ........................................................................................................................................... 67

4.5.2 IEEE 802.1p/Q............................................................................................................................... 68

4.5.3 RTCP Support ................................................................................................................................. 69

4.6 Configuration Files...................................................70

4.6.1 SIP Configuration - sip.cfg .............................................................................................................. 70

4.6.1.1 Protocol <volpProt/>...........................................................................................................71

4.6.1.2 Dial Plan <dialplan/>..........................................................................................................77

4.6.1.3 Localization <localization/> .................................................................................................78

4.6.1.4 User Preferences <user_preferences/> ..................................................................................82

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

4.6.1.5 Tones <tones/> .................................................................................................................83

4.6.1.6 Sampled Audio for Sound Effects <sampled_audio/> ..............................................................85

4.6.1.7 Sound Effects <sound_effects/>...........................................................................................87

4.6.1.8 Voice Settings <voice/> ......................................................................................................92

4.6.1.9 Quality of Service <QOS/> ..................................................................................................102

4.6.1.10 Basic TCP/IP <TCP_IP/>...................................................................................................104

4.6.1.11 Web Server <HTTPD/>......................................................................................................107

4.6.1.12 Call Handling Configuration <call/>.....................................................................................108

4.6.1.13 Directory <directory/>.......................................................................................................110

4.6.1.14 Fonts <font/>..................................................................................................................111

4.6.1.15 Keys <keys/>..................................................................................................................113

4.6.1.16 Bitmaps <bitmaps/>.........................................................................................................115

4.6.1.17 Indicators <indicators/>...................................................................................................116

4.6.1.18 Event Logging <logging/> .................................................................................................119

4.6.1.19 Security <security/> .........................................................................................................122

4.6.1.20 Provisioning <provisioning/>..............................................................................................123

4.6.1.21 RAM Disk <RAMdisk/>......................................................................................................123

4.6.1.22 Request <request/>..........................................................................................................124

4.6.1.23 Feature <feature/>...........................................................................................................125

4.6.1.24 Resource <resource/>.......................................................................................................126

4.6.1.25 MicroBrowser <microbrowser/>..........................................................................................127

4.6.2 Per-phone Configuration - phone1.cfg............................................................................................. 128

4.6.2.1 Registration <reg/> ............................................................................................................128

4.6.2.2 Calls <call/>......................................................................................................................131

4.6.2.3 Diversion <divert/>.............................................................................................................133

4.6.2.4 Dial Plan <dialplan/>..........................................................................................................135

4.6.2.5 Messaging <msg/> ............................................................................................................137

4.6.2.6 Network Address Translation <nat/> .....................................................................................138

5 Session Initiation Protocol (SIP)........................... 141

5.1 Basic Protocols ........................................................141

5.1.1 RFC and Internet Draft Support....................................................................................................... 141

5.1.2 Request Support............................................................................................................................. 141

5.1.3 Header Support.............................................................................................................................. 142

5.1.4 Response Support........................................................................................................................... 144

5.1.4.1 1xx Responses - Provisional ..................................................................................................144

5.1.4.2 2xx Responses - Success ......................................................................................................144

5.1.4.3 3xx Responses - Redirection..................................................................................................145

5.1.4.4 4xx Responses - Request Failure............................................................................................145

5.1.4.5 5xx Responses - Server Failure..............................................................................................146

5.1.4.6 6xx Responses - Global Failure ..............................................................................................146

5.1.5 Hold Implementation...................................................................................................................... 147

5.1.6 Reliability of Provisional Responses................................................................................................. 147

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

5.1.7 Transfer......................................................................................................................................... 147

5.1.8 Third Party Call Control................................................................................................................... 147

5.2 Protocol Extensions..................................................148

5.2.1 RFC and Internet Draft Support....................................................................................................... 148

5.2.2 Request Support............................................................................................................................. 148

5.2.3 SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE).......................................... 149

5.2.4 Shared Call Appearance Signaling................................................................................................... 149

5.2.5 Bridged Line Appearance Signaling ................................................................................................. 149

6 Appendix 1 ...................................................... 151

6.1 Trusted Certificate Authority List ................................151

7 Appendix 2 ...................................................... 155

7.1 Third Party Software Attribution ................................155

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

Administrator Guide - SoundPoint® IP / SoundStation® IP Overview

1 Overview

This Administrator Guide is for the SIP 1.6.0 software release, and the bootROM 3.1.0 release.

Unless specifically described separately, the behavior and configuration of the SoundPoint® IP 301 is the same as the 300, the behavior and configuration of the SoundPoint® IP 501 is the same as the 500, the behavior and configuration of the SoundPoint® IP 601 is the same as the 600.

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-

Administrator Guide - SoundPoint® IP / SoundStation® IP Overview

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 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.

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.

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.

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

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 11.

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 2.2.1.3 Local User Interface Setup Menus on page 7.

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 2.2.1.3.2 DHCP Menu

on page 8

42 then 4

•- •

•• •

•- •

address

DNS domain

•- •

Configuration File (Phase 2: application only)

 priority when more than one source exists 

Local FLASH

See 2.2.1.3.2

VLAN ID

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

DHCP Menu

on page 8

Special Case: Cisco Discovery Protocol (CDP)

rides Local FLASH which overrides DHCP VLAN

Discovery.

over-

2.2.1.2 Provisioning File Transfer

The bootROM on the phone performs the provisioning functions of downloading the bootROM, the <Ethernet address>.cfg file, and the SIP application and uploading log files. The SIP application performs the provisioning functions of downloading all other configuration files, uploading and downloading the configuration override file and user directory, downloading the dictionary and uploading log files.

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

The protocol which will be used to transfer files from the boot server depends on several factors including the phone model and whether the bootROM or SIP application

stage of provisioning is in progress. TFTP and FTP are supported by all SoundPoint

and SoundStation Station

HTTP

IP 4000 bootROM also supports HTTP while the SIP application supports

and HTTPS. If an unsupported protocol is specified, this may result in unex-

phones. The SoundPoint® IP 301, 501, 600 and 601 and Sound-

pected behavior, see the table for details of which protocol the phone will use. The “Specified Protocol” listed in the table can be selected in the Server Type field or the Server Address can include a transfer protocol, for example http://usr:pwd@server (see 2.2.1.3.3 Server Menu on page 10). The boot server address can also be obtained via DHCP. Configuration file names in the <Ethernet address>.cfg file can include a transfer protocol, for example https://usr:pwd@server/dir/file.cfg. If a user name and password are specified as part of the server address or file name, they will be used only if the server supports them.

URL Notes: A URL should contain forward slashes instead of back slashes and should not contain spaces. Escape characters are not supported. If a user name and password are not specified, the Server User and Server Password will be used (see 2.2.1.3.3 Server Menu on page 10).

Protocol used by bootROM Protocol used by SIP Application

Specified Protocol

FTP FTP FTP FTP FTP

TFTP TFTP TFTP TFTP TFTP

HTTP FTP HTTP HTTP HTTP

HTTPS FTP HTTP Not supported. Trans-

300, 500 301, 501, 600,

601, 4000

300, 500 301, 501, 600,

fers will fail.

For downloading the bootROM and application images to the phone, the secure HTTPS protocol is not available. To guarantee software integrity, the bootROM will only download signed bootROM or application images. For HTTPS, widely recognized certificate authorities are trusted by the phone and custom certificates can be added. See 6.1 Trusted Certificate Authority List on page 151. Using HTTPS requires that SNTP be functional. Provisioning of configuration files is done by the application instead of the bootROM and this transfer can use a secure protocol.

1. HTTP is supported on all phones to download ringer wave files.

601, 4000

HTTPS

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

2.2.1.3 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 soft key to launch the main menu.

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

Navigate to Menu>Settings>Advanced>Admin Settings>Network Configuration. Advanced Settings 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

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.

2.2.1.3.1 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

DHCP Menu See 2.2.1.3.2 DHCP Menu on page 8.

, 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.

Note: Disabled when DHCP client is disabled.

Phone IP Address dotted-decimal IP

address

Subnet Mask dotted-decimal subnet

mask

IP Gateway dotted-decimal IP

address

Server Menu See 2.2.1.3.3 Server Menu on page 10.

Phone’s IP address.

Note: Disabled when DHCP client is enabled.

Phone’s subnet mask.

Note: Disabled when DHCP client is enabled.

Phone’s default router.

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

Name Possible Values

SNTP Address dotted-decimal IP

address

domain name string

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

DNS Server dotted-decimal IP

address

DNS Alternate Server dotted-decimal IP

address

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.

Description

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

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

2.2.1.3.2 DHCP Menu

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

Name

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

Possible Values

Description

DHCP Offer messages before selecting an offer.

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

Possible

Name

Values

Description

Boot Server Option 66

Custom

Static

Custom+Opt.66

Boot Server Option 128 through 254

(Cannot be the same as VLAN ID Option)

Option 66: The phone will look for option number 66 (string type) in the response received from the DHCP server. The DHCP server should send address information in option 66 which matches one of the formats described for Server Address in 2.2.1.3.3 Server Menu on page 10. If the DHCP server sends nothing then the boot server address from flash will be used.

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

2.2.1.3.3 Server Menu on page 10.

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

When the boot server parameter is set to Custom, this parameter specifies the DHCP option number in which the phone will look for its boot server.

Boot Server Option Type

VLAN Discovery Disabled No VLAN discovery via DHCP.

VLAN ID Option 128 through 254

IP Address

String

Fixed Use predefined DHCP private option values of 128,

Custom Use the number specified in the VLAN ID Option field

(Cannot be the same as Boot Server Option)

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. The IP Address must specify the boot server. The String must match one of the formats described for Server Address in 2.2.1.3.3 Server Menu on page 10

144, 157 and 191. If this is used, the VLAN ID Option field will be ignored.

as the DHCP private option value.

The DHCP private option value (when VLAN Discovery is set to Custom). Default is 129.

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

2.2.1.3.3 Server Menu

Name Possible Values Description

Server Type FTP or Trivial FTP or

HTTP or HTTPS

The protocol which the phone will use to obtain configuration and phone application files from the boot server. See 2.2.1.2 Provisioning File Transfer on page 5.

FTP = File Transfer Protocol Trivial FTP = Trivial File Transfer Protocol HTTP = Hypertext Transfer Protocol HTTPS = Hypertext Transfer Protocol, Secure

Server Address dotted-decimal IP address

domain name string

URL.

All addresses can be followed by an optional directory and optional file

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. If a URL is chosen it can include a user name and password. See 2.2.1.2 Provisioning File Transfer on page 5. All options can specify a directory and the master configuration file. See

2.2.2.1.1.1 Master Configuration Files on page 12.

Note: ":", "@", or "/" cannot be used in the user name or password.

name.

Server User any string The user name used when the phone logs into the

server if required for the selected Server Type.

Note: If the Server Address is a URL with a user name, this will be ignored.

Server Pass-

word

any string The password used when the phone logs in to the

server if required for the selected Server Type.

Note: If the Server Address is a URL with user name and password, this will be ignored.

Provisioning

Method

Provisioning String

Default or SAS-VP v2 If SAS-VP v2 is selected, provisioning is done

using XML post/response transactions.

any string The string used in XML post/response transac-

tions.

Note: Disabled when Provisioning Method is Default.

a. The server user name and password should be changed from the default values. Note that

for insecure protocols the user chosen should have very few privileges on the server.

b. Not available on SoundPoint® IP 300 and SoundPoint® IP 500 phones.

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

2.2.1.4 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 all phones except the IP

4000, simulta-

neously 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 or the phone’s SIP Configuration menu.

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.

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

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

server: bootROM image, application executable, and configuration files are all upgraded this way.

2.2.2.1.1 Configuration Files

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

2.2.2.1.1.1 Master Configuration Files

Central provisioning requires that an XML-format master configuration file be located on the boot server. Either a URL-specified master configuration file or one whose name is associated with the particular phone can be used. Refer to the following sections.

Specified Master Configuration File

The master configuration file can be explicitly specified in the boot server address, for example, http://usr:pwd@server/dir/example1.cfg. The file name must end with “.cfg” and be at least five characters long. If this file cannot be downloaded, the phone will search for the per-phone master configuration file described below.

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 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>Status>Platform>Phone menu in the application. It is also printed on a label on the back of the phone. If this file cannot be downloaded, the phone will search for the default master configuration file described below.

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.

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

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

Example:

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

Default Master SIP Configuration File -->

<!-<!--

Edit and renam e this file to < Ethe rnet-a ddress>.cfg for each

phone.

<!-- < A PPLIC ATIO N APP_FILE_PATH="sip.ld"

-->

$Revision: 1.13 $ $Date: 2004/11/26 23:30:44 $ -->

CONFIG_FILES="phone1.cfg, sip.cfg" MISC_FILES="" LOG _FILE_DIRECTO RY= ""/>

Master configuration files contain four XML attributes:

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

of 255 characters. This can be a URL with its own protocol, user name and password, for example http://usr:pwd@server/dir/sip.ld.

CONFIG_FILES A comma-separated list of configuration files. Each file name has a

maximum length of 255 characters and the list of file names has a maximum length of 2047 characters, including commas and white space. Each configuration file can be specified as a URL with its own protocol, user name and password, for example ftp://usr:pwd@server/ dir/phone2034.cfg.

MISC_FILES

A comma-separated list of other required files.

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

blank by default.

a. MISC_FILES is not normally used.

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.

2.2.2.1.1.2 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

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

Category Description Example

Application Contains parameters that affect the basic operation of the phone

such as voice codecs, gains, and tones and the IP address of an application server. All phones in an installation usually share this category of files. This file would normally be modified from Polycom templates.

User / perphone

Contains parameters unique to a particular phone user. Typical parameters include:

• display name

• unique addresses

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

sip.cfg

phone1.cfg

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.

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

2.2.2.1.2 Deploying a Boot Server for the Phones

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

server for SoundPoint

Step: Instructions:

IP and SoundStation® IP phones.

1. Set up boot server:

Note: Typically all phones are configured with the same server account, but the server 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.

2. Copy all files: Copy all files from the distribution zip file to the

3. Create per-phone configuration

files

Install boot server application or locate suitable existing server. Use RFC-compliant servers.

Create account and home directory. phone may open multiple connections to the server.

The phone will attempt to upload log files, a configuration override file, and a directory file to the server. This requires that the phone’s account has delete, write, and read permissions. The phone will still function without these permissions but will not be able to upload files.

The files downloaded from the server by the phone should be made read-only.

phone home directory. Maintain the same folder hierarchy.

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.

Note that each

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.)

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

Step: Instructions:

4. Edit sip.cfg: See 4.6 Configuration Files on page 70, particularly

for SIP server address.

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

5. Decide on boot server security pol-

icy:

Polycom recommends allowing file uploads to the boot 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 2.2.2.1.1.1 Master Configuration Files on page 12).

File permissions should give the minimum access required, and the account used should have no other rights on the server.

The phone's server 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.

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

Step: Instructions:

6. Reboot phones after configuring

their boot server via DHCP or statically:

a. If the provisioning protocol requires an account name and password, the server account

name and password must match those configured in the phones. Defaults are: provisioning protocol: FTP, name: PlcmSpIp, password: PlcmSpIp

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

See 2.2.1 Basic Network Setup on page 4.

To reboot phones, 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 & IP 301: Volume-, Volume+, Hold and Do Not Disturb

IP 500 & IP 501: Volume-, Volume+, Hold, and Messages

IP 600 & IP 601: 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!

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 sip.cfg. For more information, see 4.6.1.11 Web Server <HTTPD/> on page 107. The phone’s local user interface also permits

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

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:

Administrator password required.

User password required. Restart Phone

Network Configuration SIP Configuration SSL Security settings Reset to Default - local configuration, device settings, and file system format

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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

3 Features

This section describes the many features and corresponding administration points of

SoundPoint uration Files on page 71.

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: sip.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 4.6.1.23 Feature <feature/> on page 125.

None.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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: sip.cfg

Web S e rver (if enabled)

Local Telephone User Interface

Specify whether or not to use directory name substitution.

• For more information, see 4.6.1.4 User Preferences <user_preferences/> on page 82.

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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Central (boot server)

Local

Configuration file: sip.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, see 4.6.1.23 Feature <feature/> on page 125.

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

• For more information, see 4.6.2.2.3 Missed Call Configuration <serverMissedCall/> on page 132.

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: sip.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 4.6.1.15 Keys <keys/> on page 113.

None.

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

IP 300, IP 301, IP 500, IP 501, IP 600, IP 601 and SoundStation® IP

4000 models.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

SoundPoint IP 300 and IP 301 SIP Key Layout

Sel

GHI

PQRS

ABC21DEF

ABC

JKL

TUV87WXYZ

TUV

OPER

DEF

MNOJKLGHI

MNO

WXYZ

Del

Do Not Disturb

Redial

Hold

Key ID

SoundPoint IP 500 and IP 501 SIP Key Layout

Conference

Directories

Services

Call Lists

Conference

Transfer

Redial

19 20 21

GHI

PQRS

ABC21DEF

ABC

JKL

TUV87WXYZ

TUV

OPER

DEF

MNOJKLGHI

MNO

WXYZ

222324

Sel

Del

Messages

Do Not Disturb

Hold

Key ID

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

SoundPoint IP 600 and IP 601 SIP Key Layout

Directories

Services

Conference

Transfer

Redial

Hold

ABC21DEF

ABC

19 20 21

JKL

PQRS

GHI

TUV87WXYZ

TUV

OPER

MNO

WXYZ

25262728

222324

DEF

MNOJKLGHI

131415

SoundStation IP 4000 SIP Key Layout

Key ID

Sel

Del

Messages

Do Not Disturb

Hold

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Key ID IP 300 & 301

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

IP 500 & 501 Function

IP 600 & 601 Function

IP 4000 Function

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

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Key ID IP 300 & 301

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

IP 500 & 501 Function

IP 600 & 601 Function

IP 4000 Function

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 4.6.1.8.2 Volume Persistence <volume/> on page 94.

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

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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 L16/160002 (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: sip.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:

• 4.6.1.3.3 Call Progress Tones <callProgTones> on page 81,

• 4.6.1.6 Sampled Audio for Sound Effects <sampled_audio/> on page 85,

• 4.6.1.7 Sound Effects <sound_effects/> on page 87.

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

2. L16/16000 is not supported on SoundPoint® IP 300, 301 and SoundStation® IP 4000 phones.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

distinctive treatment include the calling party name or SIP contact (number or URL format).

Administration: Distinctive Incoming Call Treatment

For more information, see 3.1.17 Local Contact Directory on page 29.

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 26. 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

Configuration file: phone1.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 4.6.1.1.3.2 Alert Information <alertInfo/> on page 74.

Specify the ring type to be used for each line.

• For more information, see 4.6.2.1 Registration <reg/ > on page 128.

This file can be created manually using an XML editor.

• For more information, see 3.1.17.1 Local Contact Directory File Format on page 31.

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.

3.1.14 Distinctive Call Waiting

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

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Central (boot server)

Local

Configuration file: sip.cfg

Web S erv e r (if enabled)

Local Phone User Interface

3.1.15 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: sip.cfg

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

• For more information, see 4.6.1.1.3.2 Alert Information <alertInfo/> on page 74.

None.

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

• For more information, see 4.6.1.12 Call Handling

Central (boot server)

Local

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

Configuration <call/> on page 108.

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

• For more information, see 4.6.2.2.1 Do Not Disturb <donotdisturb/> on page 131.

None.

Enable or disable DND using the “Do Not Disturb” key on the SoundPoint IP 300, 301, 500, 501 and 600 or the Features menu on the SoundStation IP 4000.

3.1.16 Handset, Headset, and Speakerphone

SoundPoint® IP phones come standard with a handset and a dedicated connector is provided for a headset (not supplied). The SoundPoint® IP 500, 501, 600 and 601 phones have full-duplex speakerphones. The SoundPoint have a listen-only speakerphone. The SoundPoint

phones provide dedicated keys for

IP 300 and 301 phones

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

convenient selection of either the speakerphone or headset. The SoundStation® IP 4000 phones are full-duplex speakerphones.

Central (boot server)

Local

Configuration file: sip.cfg

Web Server (if enabled)

Local Phone User Interface

Enable or disable persistent headset mode.

• For more information, see 4.6.1.4 User Preferences

Enable or disable persistent headset mode.

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

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.

3.1.17 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.

<user_preferences/> on page 82.

See the following table for further information.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Central (boot server)

Configuration file: sip.cfg

XML file: 000000000000-directory.xml

XML file: <Ethernet address>-directory.xml

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

• For more information, see 4.6.1.13 Directory <directory/> on page 110.

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 3.1.17.1 Local Contact Directory File Format on page 31.

This file can be created manually using an XML editor.

• For information on file format, see 3.1.17.1 Local Contact Directory File Format on page 31.

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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

3.1.17.1 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>

Element Permitted Values Interpretation

fn UTF-8 encoded string of up to

40 bytes

ln UTF-8 encoded string of up to

40 bytes

first name

last name

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Element Permitted Values Interpretation

ct UTF-8 encoded string contain-

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

sd Null, 1 to 40 speed-dial index

rt Null, 1 to 21 ring type

dc UTF-8 encoded string contain-

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

ad 0,1 auto divert

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.

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

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.

divert contact

The forward-to address for the autodivert feature.

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

ar 0,1

bw 0,1 buddywatching

bb 0,1 buddyblock

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.

3.1.18 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 Send 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

auto-reject

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

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

If 1, block this contact from watching this phone.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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.

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 4.6.1.2 Dial Plan <dialplan/> on page 77.

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 4.6.2.4 Dial Plan <dialplan/> on page 135.

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

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

None.

3.1.19 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.

3.1.20 Multiple Line Keys per Registration

More than one line key can be allocated to a single registration (phone number or line). The number of line keys allocated per registration is configurable.

Central (boot server)

Configuration file: phone1.cfg

Specify the number of line keys to assign per registration.

• For more information, see 4.6.2.1 Registration <reg/> on page 128.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Web S erve r (if enabled)

Local

Local Phone User Interface

Specify the number of line keys to assign per registration. 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 the number of line keys to assign per registration 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 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. The number of concurrent calls per line key is configurable. Each registration can have more than one line key assigned to it, see 3.1.20 Multiple Line Keys per Registration on page 33.

Central (boot server)

Configuration file: sip.cfg

Configuration file: phone1.cfg

Specify the default number of calls which can be active or on hold per line key.

• For more information, see 4.6.1.12 Call Handling Configuration <call/> on page 108.

Specify per-registration the number of calls which can be active or on hold per line key assigned to that registration. This will override the default value specified in sip.cfg.

• For more information, see 4.6.2.1 Registration <reg/> on page 128.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Web S erve r (if enabled)

Local

Local Phone User Interface

Specify the default number of calls which can be active or on hold per line key and the number of calls per registration which can be active or on hold per line key assigned to that registration. Navigate to:

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

Specify per-registration the number of calls which can be active or on hold per line key assigned to that registration 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.

3.1.22 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 149.

See the following table for further information.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Central (boot server)

Configuration file: sip.cfg

Configuration file: phone1.cfg

Specify whether diversion should be disabled on shared lines.

• For more information, see 4.6.1.12.1 Shared Calls <shared/> on page 109.

Specify line-seize subscription period.

• For more information, see 4.6.1.1.2 Server <server/> on page 71.

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

• For more information, see 4.6.1.1.3.4 Special Events <specialEvent/> on page 76.

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 4.6.2.1 Registration <reg/> on page 128.

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

• For more information, see 4.6.2.3 Diversion <divert/> on page 133.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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

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.

3.1.23 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 function. For more information, see 5.2.5 Bridged Line Appearance Signaling on page 149.

See the following table for further information.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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

Central (boot server)

Local

Configuration file: sip.cfg

Configuration file: phone1.cfg

Web S erve r (if enabled)

Specify whether diversion should be disabled on shared lines.

• For more information, see 4.6.1.12 Call Handling Configuration <call/> on page 108.

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 4.6.2.1 Registration <reg/> on page 128.

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

• For more information, see 4.6.2.3 Diversion <divert/> on page 133.

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

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.

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

See the following table for further information.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Configuration File: sip.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 4.6.1.14 Fonts <font/> on page 111.

Specify which bitmaps to use.

• For more information, see 4.6.1.16 Bitmaps <bitmaps/> on page 115.

Specify how to create animations and LED indicator patterns.

• For more information, see 4.6.1.17 Indicators <indicators/ > on page 116.

None.

3.1.25 Soft Key-Driven User Interface

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

3.1.26 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. If Presence watching is enabled for speed dial entries, their status will be shown on the idle display if the SIP server supports this feature. See 3.4.1 Presence on page 51.

See the following table for further information.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Central (boot server)

Local

XML file: <Ethernet address>-directory.xml

Web Server (if enabled) None.

Local Phone User Interface The next available Speed Dial Index is

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 3.1.17.1 Local Contact Directory File Format on page 31.

assigned to new directory entries. Key-pad short cuts are available to facilitate assigning and modifying the Speed Dial Index value for entries in the directory. The Speed Dial Index field is used to link directory entries to speed dial operations.

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.

3.1.27 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 and during a call. The clock and calendar must be synchronized to a remote SNTP timeserver. The time and date displayed on the phone will flash continuously until a successful SNTP response is

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

received to indicate that they are not accurate. The time and date display can use one of several different formats and can be turned off.

Central (boot server)

Local

Configuration file: sip.cfg

Web S erv e r (if enabled)

Local Phone User Interface

Turn time and date display on or off.

• For more information, see 4.6.1.4 User Preferences <user_preferences/> on page 82.

Set the time and date display formats.

• For more information, see 4.6.1.3.2 Date and Time <datetime/> on page 81.

Set the basic SNTP settings and daylight savings parameters.

• For more information, see 4.6.1.10.2 Time Synchronization <SNTP/> on page 104.

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 TCP/IP Setup on page 5.

The user can edit the time and date format and enable or disable the time and date display under the Settings menu.

3.1.28 Idle Display Animation

All phones except the SoundPoint® IP 300 and SoundPoint® IP 301 can display a customized animation on the idle display in addition to the time and date. For example, a company logo could be displayed.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Central (boot server)

Local

Configuration file: sip.cfg

Web S erv e r (if enabled)

Local Phone User Interface

To turn idle display animation on or off.

• For more information, see 4.6.1.17 Indicators <indicators/> on page 116.

To replace the animation used for the idle display.

• For more information, see 4.6.1.17.1 Animations <Animations/> <IP_300/>, <IP_500/>, <IP_600/> and <IP_4000/> on page 116.

To change the position of the idle display animation.

• For more information, see 4.6.1.17.4.2 Graphic Icons <gi/> <IP_300/>, <IP_500/>, <IP_600/> and <IP_4000/> on page 118.

None.

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

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 4.6.2.2.2 Automatic Offhook Call Placement <autoOffHook/> on page 131.

None.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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 signaling 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.

Central (boot server)

Local

Configuration file: sip.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 signaling is used.

• For more information, see 4.6.1.1.3 SIP <SIP/> on page 73.

Specify local hold reminder options.

• For more information, see 4.6.1.12.2 Hold, Local Reminder <hold/><localReminder/> on page 109.

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

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

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

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 three types of transfers;

• Blind transfers: The call is transferred immediately to C after A has finished dialing C’s number. User A does not hear ring-back.

• Consultation transfers which are dispatched during the proceeding state: User A dials C’s number and hears ring-back and decides to complete the transfer before C answers. This option can be disabled.

• True consultation transfers: User A dials C’s number and consults privately with C after the call is answered and then completes the transfer.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Central (boot server)

Local

Configuration file: sip.cfg

Web Server (if enabled)

Local Phone User Interface

Specify whether to allow a transfer during the proceeding state of a consultation call.

• For more information, see 4.6.1.1.3 SIP <SIP/> on page 73.

None.

3.2.4 Three-Way Conference, Local or Centralized

Local or centralized conferences3 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 4.6.1.1.3.5 Conference Setup <conference/> on page 76.

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

3. 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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

incoming call treatment feature. The user’s ability to originate calls is unaffected by all call diversion options. Each registration 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 4.6.2.3 Diversion <divert/> on page 133.

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: sip.cfg

Web S erve r (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, see 4.6.1.23 Feature <feature/> on page 125.

None.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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: sip.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: sip.cfg

Web S erve r (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, see 4.6.1.23 Feature <feature/> on page 125.

None.

Turn this feature on or off.

• For more information, see 4.6.1.23 Feature <feature/> on page 125.

None.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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: sip.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 4.6.1.23 Feature <feature/> on page 125.

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

• For more information, see 4.6.1.12 Call Handling Configuration <call/> on page 108.

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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Central (boot server)

Local

Configuration file: sip.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 4.6.1.8.1.2 Codec Profiles <profiles/> on page 93.

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: sip.cfg

Enable or disable VAD and set the detection threshold.

• For more information, see 4.6.1.8.10 Voice Activity Detection <VAD/> on page 102.

4. 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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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: sip.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 4.6.1.5.1 Dual Tone Multi-

None.

Frequency <DTMF/> on page 83.

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: sip.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 4.6.1.5.1 Dual Tone MultiFrequency <DTMF/> on page 83.

None.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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: sip.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

• 4.6.1.8.1.1 Codec Preferences <preferences/> on page 92, and

• 4.6.1.8.1.2 Codec Profiles <profiles/> on page 93.

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

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

Changes are saved to local flash and backed up to <Ethernet

Local

Local Phone User Interface

address>-phone.cfg on the boot server. Changes will permanently override global settings unless deleted via the Reset User Settings menu selection.

None.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

3.3.9 Background Noise Suppression (BNS)

This feature, designed primarily for hands-free 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 hands-free 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 hands-free operation, is used to boost the transmit gain of

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

This increases the effective user-phone

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 in the Buddies display screen or for speed dial entries on the phone’s idle display. The user can block others from monitoring her phone and is

notified when a change in monitored status occurs cast automatically to monitoring phones when the user engages in calls or invokes do-

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

6. 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 broad-

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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 3.1.17.1 Local Contact Directory File Format on page 31.

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 and 301 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 Asian languages (Chinese, Japanese, and Korean) is also included but will

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

render only on the SoundPoint® IP 600’s and 601’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. Note that within a Unicode range, some characters may not be supported due to their infrequent usage.

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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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: sip.cfg

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

For more information, see:

• 4.6.1.3.1 Multilingual <multilingual/> on page 79, and

• 4.6.1.3.1.1 Adding New Languages on page 80.

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 4.6.1.14 Fonts <font/> on page 111.

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-

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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: sip.cfg

Web Server (if enabled)

Local Phone User Interface

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

• For more information, see 4.6.1.5.2 Chord Sets <chord_sets/> on page 84 and 4.6.1.3.3 Call Progress Tones <callProgTones> on page 81.

Specify downloaded sampled audio files for advanced call progress tones.

• For more information, see 4.6.1.6 Sampled Audio for Sound Effects <sampled_audio/> on page 85.

Specify patterns.

For more information, see:

• 4.6.1.7.1 Patterns <patterns/> on page 87, and

• 4.6.1.7.1.1 Call Progress Patterns on page 89.

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 SoundPoint

IP 500, 501, 600

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

and 601). A message-waiting signal from a voicemail server will trigger the messagewaiting indicator to flash.

Central (boot server)

Configuration file: sip.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 4.6.1.4 User Preferences <user_preferences/> on page 82.

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 SoundPoint sages 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 4.6.2.5 Messaging <msg/> on page 137.

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 SoundPoint sages are still accessible from the Main Menu).

IP 500, 501, 600 and 601 (instant mes-

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

On a per-registration basis, specify a subscribe contact for

Local

Local Phone User Interface

solicited NOTIFY applications, a callback mode (self callback 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 and 301 support a maximum of two registrations, the SoundPoint the SoundPoint

IP 600 and 601 support six. With the attachment of one or more

IP 500 and 501 support three and

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Expansion Modules, the SoundPoint® IP 601 supports an additional six registrations. A maximum of three Expansion Modules can be attached.

Each registration can be mapped to one or more line keys (a line key can be used for only one registration). The user can select which registration to use for outgoing calls or which 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 4.6.1.1.1 Local <local/> on page 71 and 4.6.1.1.2 Server <server/> on page 71.

For up to twelve registrations, specify a display name, a SIP address, an optional display label, an authentication user ID and password, the number of line keys to use, 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 4.6.2.1 Registration <reg/> on page 128.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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#se

For up to six registrations (depending on the phone model, in this case the maximum is six even for the IP 601), specify a display name, a SIP address, an optional display label, an authentication user ID and password, the number of line keys to use, 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, a default SIP server to register to and registration information for up to twelve registrations (depending on the phone model). 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

4.6.1.1.1 Local <local/> on page 71, 4.6.1.1.2 Server <server/> on page 71 and 4.6.2.1 Registration <reg/> on page 128.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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)

Local

Configuration file: sip.cfg

Configuration file: phone1.cfg

Web S erve r (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, see 4.6.1.23 Feature <feature/> on page 125.

Enable this feature per registration.

• For more information, see 4.6.2.1 Registration <reg/> on page 128.

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)

Configuration file: sip.cfg

Configuration file: phone1.cfg

Turn this feature on or off.

• For more information, see 4.6.1.23 Feature <feature/> on page 125.

Enable this feature per registration.

• For more information, see 4.6.2.1 Registration <reg/> on page 128.

Web S erve r (if enabled)

Local

Local Phone User Interface

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

None.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

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

4.6.1.1.2 Server <server/> on page 71 and 4.6.2.1 Registration <reg/> on page 128.

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)

Configuration file: sip.cfg

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

• For more information, see 4.6.1.25 MicroBrowser <microbrowser/> on page 127.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

Web Server (if enabled)

Local

Local Phone User Interface

Specify the Services browser home page and proxy to use.

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

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 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: sip.cfg

Web Server (if enabled)

Local Phone User Interface

3.8.2 Custom Certificates

When trying to establish a connection to a boot server for application provisioning, the phone trusts certificates issued by widely recognized certificate authorities. See 6.1 Trusted Certificate Authority List on page 151. In addition, custom certificates can be added to the phone. This is done by using the SSL Security menu on the phone to pro-

Specify the minimum lengths for the user and administrator passwords.

• For more information, see 4.6.1.19.1 Password Lengths <pwd/><length/> on page 122.

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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

vide the URL of the custom certificate then select an option to use this custom certificate.

Central (boot server)

Local

Configuration file: None.

Web Server (if enabled)

Local Phone User Interface

None.

The custom certificate can be specified and the type of certificate to trust can be set under the Settings menu.

3.8.3 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)

Configuration File: sip.cfg

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

• For more information, see 4.6.1.1.3.3 Request Validation <requestValidation/> on page 75.

Web Server (if enabled)

Local

Local Phone User Interface

None.

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

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 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 4.6.1.9 Quality of Service <QOS/> on page 102.

IP phones should be configured to give voice traffic emanating from the

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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

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: sip.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 4.6.1.10.3.1 RTP <RTP/> on page 106.

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>/netConf.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.

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

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 4.6.2.6 Network Address Translation <nat/> on page 138.

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

Navigate to: http://<phoneIPAddress>/netConf.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

Notes which accompany the software. Changes to site-wide configuration files such as sip.cfg can be done manually, but a scripting tool is useful to change per-phone config

uration files.

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

4. Reboot the phones. See Manual Reboot: Menu Option or Key Presses on page 65.

For more information, see 2.2.2 Application Configuration on page 11.

Manual Reboot: Menu Option or Key Presses

To reboot phones manually, a menu option can be selected or a key combination can be used. The menu option is called Restart Phone and it is found in the Settings menu.

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

For the key combination, press and hold the following keys simultaneously until a confirmation tone is heard or for about three seconds:

SoundPoint® IP 300 and 301:

SoundPoint

SoundStation

Centralized Reboot

IP 500 and 501:

IP 600 and 601:

IP 4000:

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

4.6.1.1.3.4 Special Events <specialEvent/> on page 76.

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 123.

4.4 Event Logging

Volume-, Volume+, Hold, Do Not Disturb

Volume-, Volume+, Hold, Messages

Volume-, Volume+, Mute, Messages

*, #, Volume+, Select

4.6.1.20 Provisioning <provisioning/> on

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-config-

urable 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.

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

7. Note: HTTP and TFTP don’t support append mode unless server settings are changed for this.

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

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:

SoundPoint® IP 300 and 301:

SoundPoint

SoundStation

IP 500 and 501:

IP 600 and 601:

IP 4000:

Line1, Line2, Arrow Up, Arrow Down

The four arrow keys.

Menu, Exit, Off-hook/Hands-free, 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: 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 4.6.1.18 Event Logging <logging/> on page 119.

Specify different directory to use for log files if desired.

• For more information, see 2.2.2.1.1.1 Master Configuration Files on page 12.

Specify a multitude of event logging settings.

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

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

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

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: sip.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:

1. when it has a valid VLAN ID set in its network configuration, or

2. is instructed to tag packets via Cisco Discovery Protocol (CDP) running on a connected Ethernet switch, or

3. a VLAN ID is obtained from DHCP (see 2.2.1.3.2 DHCP Menu on page 8).

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 4.6.1.9.2 IP TOS <IP/> on page 103.

Specify IP TOS settings.

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

None.

Central (boot server)

Local

Configuration file: sip.cfg

Web S erv e r (if enabled)

Local Phone User Interface

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

• For more information, see 4.6.1.9.1 Ethernet IEEE

802.1p/Q <Ethernet/> on page 102.

Specify 802.1p/Q settings.

Navigate to http://<phoneIPAddress>/netConf.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>Advanced>Admin Settings>Network Configuration

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

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

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.

Administrator Guide - SoundPoint IP / SoundStation® IP Optimization

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 two sections:

• 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 following table.

Special Character Required Character Sequence in Text Editor

” "

’ '

4.6.1 SIP Configuration - sip.cfg

The configuration file sip.cfg contains SIP protocol and core configuration settings that would typically apply to an entire installation and must be set before the phones will be operational, unless changed via the local web server interface or local menu settings on the phone. Settings include the local port used for SIP signaling, the address and ports of a cluster of SIP servers, and other parameters. The following sections describe each of these parameters.

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

4.6.1.1 Protocol <volpProt/>

4.6.1.1.1 Local <local/>

Permitted

Attribute

volpProt.local.port 0 to 65535 5060 Local port for sending and receiving SIP signaling

4.6.1.1.2 Server <server/>

Values

Default Interpretation

packets.

If set to 0 or Null, 5060 is used for the local port but it is not advertised in the SIP signaling.

If set to some other value, that value is used for the local port and it is advertised in the SIP signaling.

Permitted

Attribute

voIpProt.server.x.address dotted-decimal

voIpProt.server.x.port 0, Null, 1 to

Values

IP address or host name

65535

Default Interpretation

Null IP address or host name and

port of a SIP server that accepts registrations. Multiple servers

Null

can be listed starting with x=1, 2, ... for fault tolerance.

If port is 0 or Null: If voIpProt.server.x.address is a hostname and voIpProt.server.x.transport is set to DNSnaptr, do NAPTR then SRV lookups.

If voIpProt.server.x.transport is set to TCPpreferred or UDPonly then use 5060 and don’t advertise the port number in signalling.

If voIpProt.server.x.address is an IP address, there is no DNS lookup and 5060 is used for the port but it is not advertised in signaling.

If port is 1 to 65535: This value is used and it is advertised in signaling.

Administrator Guide - SoundPoint IP / SoundStation® IP Optimization

Permitted

Attribute

Values

Default Interpretation

voIpProt.server.x.transport DNSnaptr or

TCPpreferred or UDPonly

voIpProt.server.x.expires positive integer,

minimum 300

voIpProt.server.x.register 0, 1 1 If set to 0, calls can be routed to

DNSna ptr

3600 Requested registration period

If set to Null or DNSnaptr: If voIpProt.server.x.address is a hostname and voIpProt.server.x.port is 0 or Null, do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC

3263. If voIpProt.server.x.address is an IP address, or a port is given, then UDP is used.

If set to TCPpreferred: TCP is the preferred transport, UDP is used if TCP fails.

If set to UDPonly: Only UDP will be used.

in seconds

an outbound proxy without registration.

voIpProt.server.x.retryTimeOut Null or

non-negative integer

voIpProt.server.x.retryMaxCount Null or

non-negative integer

voIpProt.server.x.expires.lineSeize positive integer,

minimum 10

a. This is the phone’s requested registration period. The period negotiated with the server may

be different. The phone will attempt to re-register when half the negotiated period has expired.

0 If set to 0 or Null, use standard

RFC 3261 signaling retry behavior. Otherwise retryTimeOut determines how often retries will be sent.

Units = milliSeconds. (Finest resolution = 100ms).

3 If set to 0 or Null, 3 is used.

retryMaxCount retries will be attempted before moving on to the next available server.

30 Requested line-seize subscrip-

tion period.

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

4.6.1.1.3 SIP <SIP/>

Permitted

Attribute

voIpProt.SIP.useRFC2543hold 0, 1 0 If set to 1, use the obsolete c=0.0.0.0

voIpProt.SIP.lcs 0, 1 0 If set to 1, the proprietary “epid”

voIpProt.SIP.sendCompactHdrs 0, 1 0 If set to 0, SIP header names generated

Values

Default Interpretation

RFC2543 technique, otherwise, use SDP media direction attributes (such as a=sendonly) per RFC 3264 when initiating hold. In either case, the phone processes incoming hold signaling in either format.

parameter is added to the From field of all requests to support Windows Live Communications Server.

by the phone use the long form, for example ‘From’.

voIpProt.SIP.WM50 0, 1 0

voIpProt.SIP.keepalive.sessionTimers

voIpProt.SIP.requestURI.E164.addGlobalPrefix

voIpProt.SIP.allowTransferOnProceeding

0, 1 0 If set to 1, the session timer will be

0, 1 0 If set to 1, ‘+’ global prefix is added to

0, 1 1 If set to 1, a transfer can be completed

If set to 1, SIP header names generated by the phone use the short form, for example ‘f’.

If set to 1, Windows Messenger will be supported.

If set to 0, Windows Messenger will be supported.

enabled.

If set to 0, the session timer will be disabled, and the phone will not declare “timer” in “Support” header in INVITE. The phone will still respond to a re-INVITE or UPDATE. The phone will not try to re-INVITE or do UPDATE even if remote endpoint asks for it.

E.164 user parts in sip: URIs:.

during the proceeding state of a consultation call. This is the default.

5.0

4.7

If set to 0, a transfer is not allowed during the proceeding state of a consultation call.

Administrator Guide - SoundPoint IP / SoundStation® IP Optimization

4.6.1.1.3.1 Outbound Proxy <outboundProxy/>

Permitted

Attribute

Values

Default Interpretation

voIpProt.SIP.outboundProxy.address dotted-deci-

mal IP address or host name

voIpProt.SIP.outboundProxy.port 1 to 65535 5060

voIpProt.SIP.outboundProxy.transport DNSnaptr or

TCPpreferred or UDPonly

Null IP address or host name and

DNSnaptrIf set to Null or DNSnaptr:

port of a SIP server to which the phone shall send all requests.

If voIpProt.SIP.outboundProxy.address is a hostname and voIpProt.SIP.outboundProxy.port is 0 or Null, do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If voIpProt.SIP.outboundProxy.address is an IP address, or a port is given, then UDP is used.

If set to TCPpreferred: TCP is the preferred transport, UDP is used if TCP fails.

If set to UDPonly: Only UDP will be used.

4.6.1.1.3.2 Alert Information <alertInfo/>

Permitted

Attribute

volpProt.SIP.alertInfo.x.value string to com-

voIpProt.SIP.alertInfo.x.class positive

Values

pare against the value of Alert-Info headers in INVITE requests

integer

Default Interpretation

Null Alert-Info fields from

INVITE requests will be compared against as many of these parameters as are specified (x=1, 2, ..., N) and if a match is found, the behavior described in the correspond-

Null

ing ring class (see 4.6.1.7.2 Ring type <ringType/> on page 91) will be applied.

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

4.6.1.1.3.3 Request Validation <requestValidation/>

Attribute Permitted Values Default Interpretation

voIpProt.SIP.requestValidation.x.request

voIpProt.SIP.requestValidation.x.method

One of: “INVITE”, “ACK”, “BYE”, “REGISTER”, “CANCEL”, “OPTIONS”, “INFO”, “MESSAGE”, “SUBSCRIBE”, “NOTIFY”, “REFER”, “PRACK”, or “UPDATE”

Null or one of: “source”, “digest” or “both”/”all”

Null Sets the name of the method

for which validation will be

applied

Null If Null, no validation is done.

Otherwise this sets the type of validation performed for the request:

source: ensure request is received from an IP address of a server belonging to the set of target registration servers;

digest: challenge requests with digest authentication using the local credentials for the associated registration (line);

both or all: apply both of the above methods

voIpProt.SIP.requestValidation.x.request.y.event

A valid string Null Determines which events

specified with the Event header should be validated; only applicable when voIpProt.SIP.requestValidation.x.request is set to “SUBSCRIBE” or “NOTIFY”. If set to Null, all events will be validated.

voIpProt.SIP.requestValidation.digest.realm

A valid string PolycomSPIP Determines string used for

Realm.

a. WARNING: Intensive request validation may have a negative performance impact due to

the additional signaling required in some cases, therefore, use it judiciously.

Administrator Guide - SoundPoint IP / SoundStation® IP Optimization

4.6.1.1.3.4 Special Events <specialEvent/>

Permitted

Attribute

Values

Default Interpretation

voIpProt.SIP.specialEvent.lineSeize.nonStandard

voIpProt.SIP.specialEvent.checkSync.alwaysReboot

0, 1 1 If set to 1, process a 200 OK

0, 1 0 If set to 1, always reboot when a

4.6.1.1.3.5 Conference Setup <conference/>

Permitted Val-

Attribute

ues

response for a line-seize event SUBSCRIBE as though a lineseize NOTIFY with Subscription State: active header had been received, this speeds up processing.

NOTIFY message is received from the server with event equal to check-sync.

If set to 0, only reboot if any of the files listed in [mac].cfg have changed on the FTP server when a NOTIFY message is received from the server with event equal to check-sync.

Default Interpretation

voIpProt.SIP.conference.address

ASCII string up to 128 characters long

Null If Null, conferences are set up on the

phone locally.

If set to some value, conferences are set up by the server using the conferencing agent specified by this address. The acceptable values depend on the conferencing server implementation policy.

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

4.6.1.2 Dial Plan <dialplan/>

Permitted

Attribute

Values

Default Interpretation

dialplan.impossibleMatchHandling

dialplan.removeEndOfDial 0, 1 1 If set to 1, strip trailing # digit from

0, 1 or 2 0 If set to 0, the digits entered up to

4.6.1.2.1 Digit Map <digitmap/>

Attribute Permitted Values Default Interpretation

dialplan.digitmap string compatible with

the digit map feature of MGCP described in

2.1.5 of RFC 3435. String is limited to 512 bytes and 20 segments; a comma is also allowed; when reached in the digit map, a comma will turn dial tone back on.

and including the point where an impossible match occurred are sent to the server immediately.

If set to 1, give reorder tone.

If set to 2, allow user to accumulate digits and dispatch call manually with the

digits sent out.

Send soft key.

When this attribute is present, number-only dialing during the setup phase of new calls will be compared against the patterns therein and if a match is found, the call will be initiated automatically eliminating the need to press Send.

dialplan.digitmap.timeOut positive integer 3 Timeout in seconds for

‘T’ feature of digitmap.

4.6.1.2.2 Routing <routing/>

This configuration section allows the user to create a specific routing path for outgoing SIP calls independent of other ‘default’ configuration.

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4.6.1.2.2.1 Server <server/>

Attribute Permitted Values Default Interpretation

dialplan.routing.server.x.address

dialplan.routing.server.x.port

dotted-decimal IP address or host name

1 to 65535 5060

4.6.1.2.2.2 Emergency <emergency/>

In the following attributes, x is the index of the emergency entry description and y is the index of the server associated with emergency entry x. For each emergency entry (index x), one or more server entries (indexes (x,y)) can be configured. x and y must both use sequential numbering starting at 1.

Attribute Permitted Values Default Interpretation

dialplan.routing.emergency.x.value

Comma separated list of entries or single entry representing a SIP URL or a combination of SIP URLs.

Null IP address or host name and

port of a SIP server that will be used for routing calls. Multiple servers can be listed starting with x=1, 2, ... for fault tolerance.

Null

Example: “15,17,18”, “911”, “sos”.

This determines the URLs that should be watched for.

When one of these defined URLs is detected as having been dialed by the user, the call will automatically be directed to the defined emergency server.

dialplan.routing.emergency.x.server.y

positive integer Null Index representing the

4.6.1.3 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 4.6.1.3.3 Call Progress Tones <callProgTones> on page 81,

4.6.1.5.2 Chord Sets <chord_sets/> on page 84, and 4.6.1.7.1.1 Call Progress Patterns on page 89.

server defined in

4.6.1.2.2.1 Server <server/> on page 78 that will be used for emergency routing.

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4.6.1.3.1 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

4.6.1.3.3 Call Progress Tones <callProgTones> on page 81.

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 am/pm.

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Permitted

Attribute

Values

Interpretation

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

includes ‘D’, ‘d’ and ‘M’ and two optional commas

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

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

If attribute present, overrides lcl.datetime.date.format; D = day of week d = day M = month

Up to two commas may be included. For example: D,dM = Thursday, 3 July or Md,D = July 3, Thursday

The field may contain 0, 1 or 2 commas which can occur only between characters and only one at a time. For example: “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.

lcl.ml.lang.y.list “All” or a comma-

4.6.1.3.1.1 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 52.

separated list

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

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4.6.1.3.2 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 commas

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 commas may be included. For example: D,dM = Thursday, 3 July or Md,D = July 3, Thursday

The field may contain 0, 1 or 2 commas which can occur only between characters and only one at a time. For example: “D,,dM” is illegal.

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

above date.

4.6.1.3.3 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.

Permitted

Attribute

lcl.cpt positive

lcl.cpt.menu.x string String to specify the country or region such as

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.

Italy. 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.

Administrator Guide - SoundPoint IP / SoundStation® IP Optimization

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 4.6.1.7.1.1 Call Progress Patterns on page 89.

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.

4.6.1.4 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, hands-free 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

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

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

after a reboot.

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

Administrator Guide - SoundPoint® IP / SoundStation® IP Optimization

Permitted

Attribute

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

Values

Default Interpretation

the idle display

4.6.1.5 Tones <tones/>

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

4.6.1.5.1 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

4.6.1.5.2 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.

4.6.1.6 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

• L16/160008 (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.

8. L16/16000 is not supported on SoundPoint® IP 300, 301 and SoundStation® IP 4000 phones.

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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

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.

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4.6.1.7 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.

4.6.1.7.1 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)

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)

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Instruction Meaning Example

branch (n) Advance n instruc-

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

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)

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.

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

OR silence OR branch

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

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

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4.6.1.7.1.1 Call Progress Patterns

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

Call progress pattern number

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)

Use within phone

10 record warning

11 message waiting tone

12 alerting

13 intercom announcement tone

14 barge-in tone

4.6.1.7.1.2 Ringer Patterns

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

Ringer pattern number

2Low Trill

3 Low Double Trill

4 Medium Trill

Default description

Silent Ring

5 Medium Double Trill

6 High Trill

7 High Double Trill

8 Highest Trill

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Ringer pattern number

9 Highest Double Trill

10 Beeble

11 Triplet

12 Ringback-style

Default description

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 4.6.1.6 Sampled Audio for Sound Effects <sampled_audio/> on page 85.

Sampled audio file 2

4.6.1.7.1.3 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

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4.6.1.7.2 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.

Provide auto-answer on incoming call

Provide auto answer on incoming call after a ring period

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

Attribute Permitted Values Interpretation

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

the phone, 0 otherwise.

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

se.rt.x.type ring OR visual OR

answer OR ringanswer

interface

As defined in table above.

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.

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

4.6.1.7.1.2 Ringer Patterns on page 89.

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 89.

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.

4.6.1.7.1.1 Call

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Attribute Permitted Values Interpretation

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

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

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

4.6.1.8 Voice Settings <voice/>

4.6.1.8.1 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

4.6.1.8.1.1 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

Values

Sample Rate

Frame Size

Default Interpretation

SoundPoint 601 platforms.

1 = highest 3 = lowest Null = do not use

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

Effective Audio Bandwidth

IP 500, 501, 600 and

Polycom SOUNDPOINT IP 300, SOUNDPOINT IP 501, SOUNDPOINT IP 301 User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

1 Overview

2 Installation and Operation

2.1 Installation Models

2.2 Installation Process

2.2.1 Basic Network Setup

2.2.1.1 DHCP or Manual TCP/IP Setup

2.2.1.2 Provisioning File Transfer

2.2.1.3 Local User Interface Setup Menus

2.2.1.3.1 Main Menu

2.2.1.3.2 DHCP Menu

2.2.1.3.3 Server Menu

2.2.1.4 Reset to Factory Defaults

2.2.2 Application Configuration

2.2.2.1 Centralized Configuration

2.2.2.1.1 Configuration Files

2.2.2.1.2 Deploying a Boot Server for the Phones

2.2.2.2 Local Phone Configuration

3 Features

3.1 Basic Features

3.1.1 Call Log

3.1.2 Call Timer

3.1.3 Call Waiting

3.1.4 Called Party Identification

3.1.5 Calling Party Identification

3.1.6 Missed Call Notification

3.1.7 Configurable Feature Keys

3.1.8 Connected Party Identification

3.1.9 Context Sensitive Volume Control

3.1.10 Customizable Audio Sound Effects

3.1.11 Message Waiting Indication

3.1.12 Distinctive Incoming Call Treatment

3.1.13 Distinctive Ringing

3.1.14 Distinctive Call Waiting

3.1.15 Do-Not-Disturb

3.1.16 Handset, Headset, and Speakerphone

3.1.17 Local Contact Directory

3.1.17.1 Local Contact Directory File Format

3.1.18 Local Digit Map

3.1.19 Microphone Mute

3.1.20 Multiple Line Keys per Registration

3.1.21 Multiple Call Appearances

3.1.22 Shared Call Appearances

3.1.23 Bridged Line Appearances

3.1.24 Customizable Fonts and Indicators

3.1.25 Soft Key-Driven User Interface

3.1.26 Speed Dial

3.1.27 Time and Date Display

3.1.28 Idle Display Animation

3.2 Call Management Features

3.2.1 Automatic Off-hook Call Placement

3.2.2 Call Hold

3.2.3 Call Transfer

3.2.4 Three-Way Conference, Local or Centralized

3.2.5 Call Diversion (Call Forward)

3.2.6 Directed Call Pick-up

3.2.7 Group Call Pick-up

3.2.8 Call Park / Retrieve

3.2.9 Last Call Return

3.3 Audio Processing Features

3.3.1 Low-Delay Audio Packet Transmission

3.3.2 Jitter Buffer and Packet Error Concealment

3.3.3 Local Conference Mixing

3.3.4 Voice Activity Detection (VAD)

3.3.5 DTMF Tone Generation

3.3.6 DTMF Event RTP Payload

3.3.7 Acoustic Echo Cancellation (AEC)

3.3.8 Audio Codecs

3.3.9 Background Noise Suppression (BNS)

3.3.10 Comfort Noise Fill

3.3.11 Automatic Gain Control (AGC)

3.4 Presence and Instant Messaging Features

3.4.1 Presence

3.4.2 Instant Messaging

3.5 Localization Features

3.5.1 Multilingual User Interface

3.5.2 Downloadable Fonts