Polycom 1725-11530-200 User Manual

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Administrator’s Guide

SoundPoint®/SoundStation® IP SIP

Version 2.0

August 2006

Notices

1. Specifications subject to change without notice.

Polycom, Inc.

1565 Barber Lane, Milpitas CA 95035, USA

www.polycom.com

Part Number: 1725-11530-200 Rev A1

Administrator’s Guide - SoundPoint® IP / SoundStation® IP 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....................................................................................................................... 5

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

2.2.1.2 Provisioning File Transfer ......................................................................................................6

2.2.1.3 Local User Interface Setup Menus ..........................................................................................8

2.2.1.4 Reset to Factory Defaults......................................................................................................12

2.2.2 Application Configuration................................................................................................................ 13

2.2.2.1 Centralized Configuration......................................................................................................13

2.2.2.2 Local Phone Configuration.....................................................................................................22

2.2.3 Management of File Encryption and Decryption............................................................................... 23

2.2.3.1 Changing the Key on the Phone.............................................................................................24

3 Features .......................................................... 27

3.1 Basic Features.........................................................27

3.1.1 Call Log ......................................................................................................................................... 27

3.1.2 Call Timer ...................................................................................................................................... 27

3.1.3 Call Waiting ................................................................................................................................... 28

3.1.4 Called Party Identification............................................................................................................... 28

3.1.5 Calling Party Identification.............................................................................................................. 28

3.1.6 Missed Call Notification................................................................................................................... 28

3.1.7 Configurable Feature Keys.............................................................................................................. 29

3.1.8 Connected Party Identification ........................................................................................................ 33

3.1.9 Context Sensitive Volume Control.................................................................................................... 34

3.1.10 Customizable Audio Sound Effects................................................................................................. 34

3.1.11 Message Waiting Indication .......................................................................................................... 35

3.1.12 Distinctive Incoming Call Treatment............................................................................................... 35

3.1.13 Distinctive Ringing........................................................................................................................ 35

3.1.14 Distinctive Call Waiting ................................................................................................................. 36

3.1.15 Do-Not-Disturb............................................................................................................................. 36

3.1.16 Handset, Headset, and Speakerphone........................................................................................... 37

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

3.1.17 Local Contact Directory................................................................................................................. 38

3.1.17.1 Local Contact Directory File Format.......................................................................................39

3.1.18 Local Digit Map............................................................................................................................ 40

3.1.19 Microphone Mute ......................................................................................................................... 41

3.1.20 Multiple Line Keys per Registration ............................................................................................... 41

3.1.21 Multiple Call Appearances............................................................................................................. 42

3.1.22 Shared Call Appearances .............................................................................................................. 43

3.1.23 Bridged Line Appearances............................................................................................................. 45

3.1.24 Busy Lamp Field........................................................................................................................... 47

3.1.25 Customizable Fonts and Indicators................................................................................................. 47

3.1.26 Soft Key-Driven User Interface...................................................................................................... 48

3.1.27 Speed Dial ................................................................................................................................... 48

3.1.28 Time and Date Display.................................................................................................................. 49

3.1.29 Idle Display Animation ................................................................................................................. 51

3.2 Call Management Features........................................51

3.2.1 Automatic Off-hook Call Placement................................................................................................. 51

3.2.2 Call Hold........................................................................................................................................ 52

3.2.3 Call Transfer .................................................................................................................................. 52

3.2.4 Three-Way Conference, Local or Centralized.................................................................................... 53

3.2.5 Call Diversion (Call Forward) .......................................................................................................... 54

3.2.6 Directed Call Pick-up ...................................................................................................................... 54

3.2.7 Group Call Pick-up.......................................................................................................................... 55

3.2.8 Call Park / Retrieve ....................................................................................................................... 55

3.2.9 Last Call Return.............................................................................................................................. 56

3.3 Audio Processing Features ........................................56

3.3.1 Low-Delay Audio Packet Transmission............................................................................................. 56

3.3.2 Jitter Buffer and Packet Error Concealment ..................................................................................... 56

3.3.3 Voice Activity Detection................................................................................................................... 57

3.3.4 DTMF Tone Generation ................................................................................................................... 58

3.3.5 DTMF Event RTP Payload ................................................................................................................ 58

3.3.6 Acoustic Echo Cancellation (AEC) ..................................................................................................... 58

3.3.7 Audio Codecs.................................................................................................................................. 59

3.3.8 Background Noise Suppression (BNS).............................................................................................. 60

3.3.9 Comfort Noise Fill........................................................................................................................... 60

3.3.10 Automatic Gain Control (AGC)....................................................................................................... 60

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

3.4 Presence and Instant Messaging Features ...................60

3.4.1 Presence........................................................................................................................................ 60

3.4.2 Instant Messaging .......................................................................................................................... 61

3.5 Localization Features ...............................................61

3.5.1 Multilingual User Interface ............................................................................................................. 61

3.5.2 Downloadable Fonts ....................................................................................................................... 63

3.5.3 Synthesized Call Progress Tones...................................................................................................... 63

3.6 Advanced Server Features ........................................64

3.6.1 Voice Mail Integration..................................................................................................................... 64

3.6.2 Multiple Registrations..................................................................................................................... 66

3.6.3 ACD login / logout ......................................................................................................................... 68

3.6.4 ACD agent available / unavailable.................................................................................................. 68

3.6.5 Server Redundancy ........................................................................................................................ 68

3.6.5.1 DNS SIP Server Name Resolution...........................................................................................69

3.6.6 Microsoft® Office Live Communications Server 2005 Integration .................................................... 69

3.6.6.1 Configuration File Changes....................................................................................................71

3.7 Accessory Internet Features.......................................73

3.7.1 MicroBrowser................................................................................................................................. 73

3.8 Security Features.....................................................73

3.8.1 Local User and Administrator Privilege Levels.................................................................................. 73

3.8.2 Custom Certificates......................................................................................................................... 74

3.8.3 Incoming Signaling Validation......................................................................................................... 74

3.8.4 Configuration File Encryption.......................................................................................................... 75

4 Optimization .................................................... 77

4.1 Ethernet Switch .......................................................77

4.2 Application Network Setup .......................................77

4.2.1 Real-Time Transport Protocol Ports.................................................................................................. 77

4.2.2 Working with Network Address Translation...................................................................................... 78

4.3 Updating and Rebooting...........................................79

4.4 Event Logging .........................................................80

4.5 Audio Quality Issues and VLANs ................................81

4.5.1 IP TOS ........................................................................................................................................... 81

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

4.5.2 IEEE 802.1p/Q............................................................................................................................... 82

4.5.3 RTCP Support ................................................................................................................................. 83

4.6 Configuration Files...................................................84

4.6.1 SIP Configuration - sip.cfg .............................................................................................................. 84

4.6.1.1 Protocol <volpProt/>...........................................................................................................85

4.6.1.2 Dial Plan <dialplan/>..........................................................................................................94

4.6.1.3 Localization <localization/> .................................................................................................95

4.6.1.4 User Preferences <user_preferences/> ..................................................................................98

4.6.1.5 Tones <tones/> .................................................................................................................99

4.6.1.6 Sampled Audio for Sound Effects <sampled_audio/> ..............................................................101

4.6.1.7 Sound Effects <sound_effects/>...........................................................................................103

4.6.1.8 Voice Settings <voice/> ......................................................................................................108

4.6.1.9 Quality of Service <QOS/> ..................................................................................................118

4.6.1.10 Basic TCP/IP <TCP_IP/>...................................................................................................121

4.6.1.11 Web Server <HTTPD/>......................................................................................................125

4.6.1.12 Call Handling Configuration <call/>.....................................................................................125

4.6.1.13 Directory <directory/>.......................................................................................................128

4.6.1.14 Presence <presence/>.......................................................................................................129

4.6.1.15 Fonts <font/>..................................................................................................................129

4.6.1.16 Keys <keys/>..................................................................................................................132

4.6.1.17 Bitmaps <bitmaps/>.........................................................................................................133

4.6.1.18 Indicators <indicators/>....................................................................................................134

4.6.1.19 Event Logging <logging/> .................................................................................................137

4.6.1.20 Security <security/> .........................................................................................................140

4.6.1.21 Provisioning <provisioning/>..............................................................................................142

4.6.1.22 RAM Disk <RAMdisk/>......................................................................................................142

4.6.1.23 Request <request/>..........................................................................................................143

4.6.1.24 Feature <feature/>...........................................................................................................144

4.6.1.25 Resource <resource/>..................................................................................................

.....145

4.6.1.26 MicroBrowser <microbrowser/>..........................................................................................146

4.6.2 Per-phone Configuration - phone1.cfg............................................................................................. 149

4.6.2.1 Registration <reg/> ............................................................................................................149

4.6.2.2 Calls <call/>......................................................................................................................153

4.6.2.3 Diversion <divert/>.............................................................................................................154

4.6.2.4 Dial Plan <dialplan/>..........................................................................................................156

4.6.2.5 Messaging <msg/> ............................................................................................................159

4.6.2.6 Network Address Translation <nat/> .....................................................................................160

4.6.2.7 Attendant <attendant/> ......................................................................................................161

4.6.2.8 Roaming Buddies <roaming_buddies/>.................................................................................161

4.6.2.9 Roaming Privacy <roaming_privacy/>...................................................................................162

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

5 Session Initiation Protocol (SIP)........................... 163

5.1 Basic Protocols ........................................................163

5.1.1 RFC and Internet Draft Support....................................................................................................... 163

5.1.2 Request Support............................................................................................................................. 163

5.1.3 Header Support.............................................................................................................................. 164

5.1.4 Response Support........................................................................................................................... 166

5.1.4.1 1xx Responses - Provisional ..................................................................................................166

5.1.4.2 2xx Responses - Success ......................................................................................................166

5.1.4.3 3xx Responses - Redirection..................................................................................................167

5.1.4.4 4xx Responses - Request Failure............................................................................................167

5.1.4.5 5xx Responses - Server Failure..............................................................................................168

5.1.4.6 6xx Responses - Global Failure ..............................................................................................169

5.1.5 Hold Implementation...................................................................................................................... 169

5.1.6 Reliability of Provisional Responses................................................................................................. 169

5.1.7 Transfer......................................................................................................................................... 169

5.1.8 Third Party Call Control................................................................................................................... 169

5.2 Protocol Extensions..................................................170

5.2.1 RFC and Internet Draft Support....................................................................................................... 170

5.2.2 Request Support............................................................................................................................. 171

5.2.3 SIP for Instant Messaging and Presence Leveraging Extensions ........................................................ 171

5.2.4 Shared Call Appearance Signaling................................................................................................... 171

5.2.5 Bridged Line Appearance Signaling ................................................................................................. 172

6 Appendix 1 ...................................................... 173

6.1 Trusted Certificate Authority List ................................173

6.2 Miscellaneous Administrative Tasks ...........................175

6.2.1 Adding a Background Logo............................................................................................................. 175

7 Appendix 2 ...................................................... 177

7.1 Third Party Software Attribution ................................177

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

Administrator’s Guide’s - SoundPoint® IP / SoundStation® IP Overview

1 Overview

This Administrator Guide is for the SIP 2.0 software release and the bootROM 3.2 release.

Note

Unless specifically described separatel y, the behavior and configuration of the SoundPoint® IP 301 is the same as the 300, the behavior and configuration of the SoundPoint

the behavior and configuration of the SoundPoint

IP 601 is the same as the 600.

SoundPoint® IP and SoundStation® IP are feature-rich, enterprise-class voice communications terminals for Ethernet TCP/IP networks. They are designed to facilitate highquality audio communications. These phones are end points in the overall network topology designed to interoperate with other compatible equipment including application servers, media servers, internetworking gateways, voice bridges, and other end points.

IP 501 is the same as the 500,

Administrator’s 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 provides 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’s 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 through a system of global and perphone 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.

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

A boot server allows global and per-phone configuration to be managed centrally through 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. Multiple redundant boot servers can be configured to improve reliability.

The configuration served by the boot server can be augmented by changes made locally on the phone itself or through 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 advantages of a boot server are:

• Provides a centralized repository for application images and configuration files permits application updates and coordinated configuration parameters.

• Provides security as some parameters can only be modified using boot server configuration files.

• Provides consistency as the multilingual feature requires boot server-resident dictionary files and the customized sound effect wave files require a boot server.

• Provides common file uploads when permitted. The boot server is the repository for:

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

• local configuration changes through 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.

• Provides a common repository for the application images and configuration files. The boot server copy can be used to “repair” a damaged phone configura tion in the same way that system repair disks work for PCs.

2.2 Installation Process

Regardless of whether or not you will be installing a centrally provisioned system, the following steps are required to get your organization’s phones up and running:

1. Basic TCP/IP Network Setup such as IP address and subnet mask. For more information, refer to 2.2.1 Basic Network Setup on page 5.

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

For the detailed steps required in a boot server deployment, refer to 2.2.2.1.2 Boot Server Deployment for the Phones on page 19.

2.2.2 Application Configuration on page 13.

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

T o safeguard your files for reliability and backups, you should encrypt them. For more information, refer to 2.2.3 Management of File Encryption and Decryption on page 23.

For the latest information on system requirements, fixed problems, and workarounds, refer to the Release Notes at www.polycom.com/support/voip/ .

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 Session Initiation Protocol (SIP) phone application.

Networking starts in Phase 1. The bootROM application uses the network to query the boot server for upgrades, 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. For more information, refer to 2.2.1.1 DHCP or Manual TCP/IP Setup on page 5.

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. For more information, refer to 2.2.1.2 Provisioning File Transfer on page 6.

Basic network settings can be changed during Phase 1 using the bootROM’s setup menu. A similar menu system is present in the application for changing the same network parameters. For more information, refer to 2.2.1.3 Local User Interface Setup Menus on page 8.

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, or downloaded from configuration files. Contact Polycom Customer Support for more information on this use of configuration files. 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 or configuration file:

Parameter DHCP Option

IP address

DHCP

12 3

•- •

Configuration File (Phase 2: application only)

 priority when more than one source exists 

Local FLASH

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

Parameter DHCP Option

subnet mask

IP gateway

boot server address

SIP server address

SNTP server address

SNTP GMT offset

DNS server IP address

alternate DNS server IP

1 3

Refer to

2.2.1.3.2

DHCP Menu

on page 9

151

42 then 4

6 6

address

DNS domain

VLAN ID

Refer to

2.2.1.3.2

DHCP Menu

on page 9

DHCP

(Phase 2: application only)

•- •

Configuration File

•

•- •

Special Case: Cisco Discovery Protocol (CDP)

rides Local FLASH that overrides DHCP VLAN Dis-

covery.

Local FLASH

•

over-

a. For more information on DHCP options, go to http://www.ietf.org/rfc/rfc2132.txt?num-

ber=2132. b. This value is configurable. c. Note that the configuration file value can be configured to override the DHCP value. Refer

to tcpIpApp.sntp.address.overrideDHCP in section 4.6.1.10.2 Time Synchronization

<SNTP/> on page 121. d. Note that the configuration file value can be configured to override the DHCP value. Refer

to tcpIpApp.sntp.gmtOffset.overrideDHCP in section 4.6.1.10.2 Time Synchronization

<SNTP/> on page 121. e. This value can be obtained from a connected Ethernet switch if the switch supports CDP.

2.2.1.2 Provisioning File Transfer

The SIP application performs the provisioning functions of downloading configuration files, uploading and downloading the configuration override file and user directory, and downloading the dictionary and uploading log files.

The protocol that 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 SoundStation® phones. The SoundPoint® IP 301, 430, 501, 600 and 601 and SoundStation® IP 4000 bootROM also supports HTTP, while the SIP application sup-

and

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

ports only the mentioned platforms. If an unsupported protocol is specified, this may result in a defined behavior, see the table below for details of which protocol the phone will use. The “Specified Protocol” listed in the table can be selected in the Server T yp e field or the Server Address can include a transfer protocol, for example http:// usr:pwd@server (refer to 2.2.1.3.3 Server Menu on page 11). The boot server address can be an IP address, domain string name, or URL. The boot server address can also be obtained through 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.

Note

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 (refer to 2.2.1.3.3 Server Menu on page 11).

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, 430, 501,

600, 601, 4000

300, 500 301, 430, 501,

600, 601, 4000

HTTPS

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 cryptographically signed bootROM or application images. For HTTPS, widely recognized certificate authorities are trusted by the phone and custom certificates can be added (refer to 6.1 Trusted Certificate Authority List on page 173).

Administrator’s 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

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

main menu. Navigate to Menu>Settings>Advanced>Admin Settings>Network Configuration. Advanced Settings are locked by default. Enter the administrator password to unlock. Note that the factory default password is 456.

Phone network configuration parameters may be edited by means of:

• Main menu. Refer to 2.2.1.3.1 Main Menu on page 8.

• DHCP submenu. Refer to 2.2.1.3.2 DHCP Menu on page 9.

• Server submenu. Refer to 2.2.1.3.3 Server Menu on page 11.

• Ethernet submenu. Refer to 2.2.1.3.4 Ethernet Menu on page 12.

SETUP soft key to launch the main menu.

Use the soft keys, the arrow keys, the Sel/3, and the Del/X keys to make changes. Certain parameters 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 ob tain the

DHCP Menu Refer to 2.2.1.3.2 DHCP Menu on page 9.

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

Phone’s IP address. Note: Disabled when DHCP client is enabled.

Phone’s subnet mask. Note: Disabled when DHCP client is enabled.

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

Name Possible Values

IP Gateway dotted-decimal IP

address Server Menu Refer to 2.2.1.3.3 Server Menu on page 11. SNTP Address dotted-decimal IP

address

domain name string GMT Offset -13 through +12 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. Ethernet Refer to 2.2.1.3.4 Ethernet Menu on page 12.

EM Power

Enabled, Disabled This parameter is relevant if the phone gets

Description

Phone’s default router.

Simple Network Time Protocol (SNTP) server from which the phone will obtain the current time.

Mean Time (GMT) in half hour increments. Primary server to which the phone directs

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

Domain Name System queries.

Power over Ethernet (PoE). If enabled, the phone will set power requirements in CDP to 12W so that up to three Expansion Modules (EM) can be powered. If disabled, the phone will set power requirements in CDP to 5W which means no Expansion Modules can be powered (it will not work).

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.

b. Only available on SoundPoint

IP 601 phones.

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:

Possible Name

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

Values

Description

DHCP Offer messages before selecting an offer.

Administrator’s 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 that matches one of the formats described for Server Address in 2.2.1.3.3 Server Menu on page 11. 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. If the DHCP server sends nothing, then the boot server address from flash will be used.

Static: The phone will use the boot server configured through the Server Menu. For more information, refer to 2.2.1.3.3 Server Menu on page 11.

Custom+Opt.66: The phone will first use the custom option if present or use Option 66 if the custom option is not present. If the DHCP server sends nothing, then the boot server address from flash will be used.

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 through 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 11

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 Discov-

ery is set to Custom).

Administrator’s 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

Server Address dotted-decimal IP address

OR domain name string OR URL

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

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

The protocol that the phone will use to obtain configuration and phone application files from the boot server. Refer to 2.2.1.2 Provisioning File Transfer on page 6.

The boot server to use if the DHCP client is disabled, the DHCP server does not send a boot server option, or the Boot Server parameter is set to Static. The phone can contact multiple IP addresses per DNS name. These redundant boot servers must all use the same protocol. If a URL is used it can include a user name and password. Refer to 2.2.1.2 Provisioning File Transfer on page 6. A directory and the master configuration file can be specified.

Note: ":", "@", or "/" can be used in the user name or password these characters if they are correctly escaped using the method specified in RFC 1738.

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

File Transmit Tries

Retry Wait 0 to 300

Provisioning Method

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

1 to 10 Default 3

Default 1

Default or SAS-VP If SAS-VP is selected, provisioning is done (in

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. The number of attempts to transfer a file. (An

attempt is defined as trying to download the file from all IP addresses that map to a particular domain name.)

The minimum amount of time that must elapse before retrying a file transfer, in seconds. The time is measured from the start of a transfer attempt which is defined as the set of upload/download transactions made with the IP addresses that map to a given boot server's DNS host name. If the set of transactions in an attempt is equal to or greater than the Retry Wait value, then there will be no further delay before the next attempt is started.

For more information, refer to 2.2.2.1.2 Boot Server Deployment for the Phones on page 19.

addition to the normal process).

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

Name Possible Values Description

Provisioning

String

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

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

2.2.1.3.4 Ethernet Menu

Name Possible Values Description

CDP Enabled, Disabled If enabled, the phone will use CDP. It also reports

VLAN ID Null, 0 through 4094 Phone’s 802.1Q VLAN identifier.

LAN

Auto, 10HD, 10FD, 100HD, 100FD

tions. If empty, the configured URL is used. This field is disabled when Provision ing Method

is Default.

IP 300 and SoundPoint® IP 500 phones.

power usage to the switch.

Note: Null = no VLAN tagging The network speed over the Ethernet.

The default value is Auto.

a. Only available on SoundPoint® IP 430 and 601 phones. HD means half duplex and FD

means full duplex.

Auto, 10HD, 10FD, 100HD, 100FD

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, do one of the following during the countdown process in the bootROM:

• On all phones except the IP 430 and 4000, simultaneously press and hold the 4, 6, 8 and * dial pad keys until the password prompt appears.

• On the IP 430, simultaneously press and hold the 1, 3, 5 and 7 dial pad keys until the password prompt appears.

• 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. Resetting to factory defaults will also reset the administrator password (factory default password is 456).

The network speed over the Ethernet. The default value is Auto.

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

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 optimize your system.

The following sections discuss the available configuration options:

• Centrally provisioned configuration. Refer to 2.2.2.1 Centralized Configuration on page 13.

• Local phone-based configuration. Refer to 2.2.2.2 Local Phone Configuration on page 22.

2.2.2.1 Centralized Configuration

A boot server allows global and per-phone configuration to be managed centrally through XML-format configuration files that are downloaded by the phones at boot time. 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 cache with the boot 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 file 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.

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 upper or lower case, for example, 0004f200106c.cfg. The Ethernet address can be viewed using the ABOUT soft key during the auto-boot countdown of the bootROM or through the Menu>Status>Platform>Phone menu in the application. It is also printed

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

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.

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

<?xml version=”1.0” standalone=”yes”?>    < APPLICATION APP_FILE_PATH=”sip.ld” CONFIG_FILES=”phone1.cfg, sip.cfg” MISC_FILES=”” LOG FILE DIRECTORY=”” OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY=""/>

Master configuration files contain six XML attributes:

APP_FILE_PATH The path name of the application executable. It can have 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. Dictionary resource

files listed here will be stored in the phone's flash file system. So if the phone reboots at a time when the boot server is unavailable, it will still be able to load the preferred language.

Note: On the IP 500, there is insufficient room for a language file. Specifying one will cause a reboot loop.

LOG_FILE_DIRECTORY An alternative directory to use for log files if required. A URl can also

be specified. This is blank by default.

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

CONTACTS_DIRECTORYAn alternative directory to use for user directory files if required. A

URl can also be specified. This is blank by default.

OVERRIDES_DIRECTORYAn alternative directory to use for configuration overrides files if

required. A URl can also be specified. This is blank by default.

Important

Be aware of the limited permanent storage on the phone(s).

Important

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 changing the baseline configuration files for an entire system.

For more information, refer to the “Configuration File Management on SoundPoint whitepaper at www.polycom.com/support/voip/.

IP Phones”

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.

Site-specific settings  Refer to the “Configuration File Management on SoundPoint® IP Phones” whitepaper at

www.polycom.com/support/voip/ . Per-phone settings  phoneXXXX.cfg Application settings  sip.cfg

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. Polycom recommends that you create another file with your organization’s modifications. If you must change any Polycom templates, back them up first.

sip.cfg

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

Category Description Example

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 ver-

sion of user files derived from Polycom templates.

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. The configuration files which accompany a specific release of the SIP software must be used together with that software. Failure to do this may render the phone unusable.

2.2.2.1.1.3 Setting Flash Parameters from Configuration Files

Any field in the bootROM setup menu and the application SIP Configuration menu can be set through a configuration file.

phone1.cfg

A DHCP server can be configured to point the phones to a boot server that has the required configuration files. The new settings will be downloaded by the phones and used to configure them. This removes the need for manual interaction with phones to configure basic settings. This is especially useful for initial installation of multiple phones.

These device settings are detected when the application starts. If the new settings would normally cause a reboot if they were changed in the application Network Configuration menu then they will cause a reboot when the application starts.

Important

The parameters for this feature should be put in separate configuration files to simplify maintenance. Do not add them to existing configuration files (such as sip.cfg). One new configuration file will be required for parameters that should apply to all phones, and individual configuration files will be required for phone-specific parameters such as SIP registration information.

The global device.set parameter must be enabled when the initial installation is done, and then it should be disabled. This prevents subsequent reboots by individual phones triggering a reset of parameters on the phone that may have been tweaked since the initial installation.

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

Important

This feature is very powerful and should be used with caution. For example, an incorrect setting could set the IP Address of multiple phones to the same value.

Note that some parameters may be ignored, for example if DHCP is enabled it will still override the value set with device.net.ipAddress.

Individual parameters are checked to see whether they are in range, however, the interaction between parameters is not checked. If a parameter is out of range, an error message will appear in the log file and parameter will not be used.

Incorrect configuration could cause phones to get into a reboot loop. For example, server A has a configuration file that specifies that server B should be used, which has a configuration file that specifies that server A should be used.

Polycom recommends that you test the new configuration files on two phones before initializing all phones. This should detect any errors including IP address conflicts.

Name Possible Values Description

device.set 0 or 1

default = 0

device.xxx.yyy.set 0 or 1

default = 0

device.net.ipAddress dotted-decimal IP address Phone's IP address.

device.net.subnetMask dotted-decimal IP address Phone's subnet mask.

device.net.IPgateway dotted-decima l IP address Phone's default router / IP gateway.

If set to 0, do not use any device.xxx.yyy fields to set any parameters. Set this to 0 after the initial installation.

If set to 1, use the device.xxx.yyy fields that have device.xxx.yyy.set = 1. Set this to 1 for the initial installation only.

If set to 0, do not use the device.xxx.yyy value. If set to 1, use the device.xxx.yyy value. For example, if device.net.ipAddress.set = 1, then

set the contents of the device.net.ipAddress field.

Note: This field is not used when DHCP client is enabled.

device.net.vlanId Null, 0 to 4094 Phone’s 802.1Q VLAN identifier.

Note: Null = no VLAN tagging

device.net.cdpEnabled 0 or 1 If set to 1, the phone will attempt to determine its

VLAN ID through the CDP.

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

Name Possible Values Description

device.dhcp.enabled 0 or 1 For description, refer to 2.2.1.1 DHCP or Manual

TCP/IP Setup on page 5.

device.dhcp.offerTimeout 1 to 600 Number of seconds the phone waits for secondary

DHCP Offer messages before selecting an offer.

device.dhcp.bootSrvUseOpt

device.dhcp.bootSrvOpt 128 to 254 (Cannot be the

device.dhcp.bootSrvOptType

device.dhcp.dhcpVlanDiscUseOpt

device.dhcp.dhcpVlanDiscOpt

device.prov.serverName any string Fo r descri ptions, refer to 2.2.1.3.3 Server Menu device.prov.serverType 0 to 4 device.prov.user any string device.prov.password any string device.prov.appProvType 0 or 1 device.prov.app-

ProvString

0 to 3 For descriptions, refer to 2.2.1.3.2 DHCP Menu

on page 9.

same as VLAN ID Option)

0 or 1

0 to 2

128 to 254 (Cannot be the same as Boot Server Option)

on page 11

any string

device.sntp.serverName any string Can be dotted-decimal IP address or domain name

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

device.sntp.gmtOffset -43200 to 46800 GMT offset in seconds, corresponding to -12 to

+13 hours.

device.dns.serverAddress dotted-decimal IP address Primary server to which the phone directs Domain

Name System queries.

device.dns.altSrvAddress dotted-decimal IP address Secondary server to which the phone directs

Domain Name System queries. device.dns.domain any string The phone’s DNS domain. device.auth.localAdmin-

Password device.auth.localUser-

Password

any string The phone’s local administrator password.

any string The phone user’s local password.

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

Name Possible Values Description

device.auth.regUserx any string The SIP registration user name for registration x

where x = 1 to 12. device.auth.regPassword any string The SIP registration password for registration x

where x = 1 to 12. device.sec.configEncryp-

tion.key

any string Configuration encryption key that is used for

encryption of configuration files.

2.2.2.1.2 Boot Server Deployment for the Phones

The following table describes the steps required for successful deployment of one or more boot servers for SoundPoint® IP and SoundStation® IP phones (except for SoundPoint® IP 300 and 500 phones). Multiple boot servers can be configured by having the boot server DNS name map to multiple IP addresses. The default number of boot servers is one and the maximum number is eight. The following protocols are supported for redundant boot servers: HTTPS, HTTP, and FTP.

All of the boot servers must be reachable by the same protocol and the content available on them must be identical. The parameters described in section 2.2.1.3.3 Server Menu on page 11 can be used to configure the number of times each server will be tried for a file transfer and also how long to wait between each attempt. The maximum number of servers to be tried is configurable. Contact Polycom Customer Support for more information.

Note

Be aware of how logs, overrides and directories are uploaded to servers that maps to multiple IP addresses. The server that these files are uploaded to may change over time.

.If you want to use redundancy for uploads, you will have to synchronize the files between servers in the background.

You may want to disable the redundancy for uploads by specifying specific IP addresses instead of URLs for logs, overrides, and directory in the MAC.cfg.

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

Step: Instructions:

1. Set up boot server(s).

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.

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

Install boot server application or locate suitable existing server(s). 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’s Guide - SoundPoint® IP / SoundStation® IP Installation and Operation

Step: Instructions:

4. Create a new configuration file (in the style of sip.cfg).

For more information on why to create another configuration file, refer to the “Configuration File Manage-

ment on SoundPoint

IP Phones” whitepaper at

www.polycom.com/support/voip/ . Refer to 4.6 Configuration Files on page 84, particu-

larly for SIP server address. Most of the default settings are typically adequate,

however, if SNTP settings are not available through 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 change 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 policy. 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 (through 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 (refer to LOG_FILE_DIRECTORY in 2.2.2.1.1.1 Master Configuration Files on page 13).

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 through file server file permissions.

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

Step: Instructions:

6. Reboot phones after configuring their

boot server through 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

Refer to 2.2.1 Basic Network Setup on page 5. 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 430, 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 cor-

rectly. 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 it is disabled through sip.cfg. For more information, refer to

4.6.1.11 Web Server <HTTPD/> on page 125. The phone’s local user interface also permits many application settings to be modified, such as SIP server address, 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.

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

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 through 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.

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

Important

Local configuration changes will continue to override the boot server-derived configuration until deleted through the Reset Local Config menu selection.

2.2.3 Management of File Encryption and Decryption

The phone can recognize encrypted files, which it downloads from the boot server and it can encrypt files before uploading them to the boot server. There must be an encryption key on the phone to perform these operations. Configuration files (excluding the master configuration file), contact directories and configuration override files can be encrypted.

A separate SDK, with a readme file, is provided to facilitate key generation and configuration file encryption and decrypt on a UNIX or Linux server. The utility is distributed as source code that runs under the UNIX operating system. A key is generated by the utility and must be downloaded to the phone so that it can decrypt the files that were encrypted on the server. The device.sec.configEncryption.key configuration file parameter is used to set the key on the phone. The utility generates a random key and

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

the encryption is Advanced Encryption Standard (AES) 128 in Cipher Block Chaining (CBC) mode. An example key would look like this:

Crypt=1;KeyDesc=companyNameKey1;Key=06a9214036b8a15b512e03d534120006;

It is recommended that all keys have unique descriptive strings in order to allow simple identification of which key was used to encrypt a file. This makes boot server management easier.

After encrypting a configuration file, it is useful to rename the file to avoid confusing it with the original version, for example rename sip.cfg to sip.enc. However, the directory and override filenames cannot be changed in this manner.

You can check whether an encrypted file is the same as an unencrypted file by:

1. Run the configFileEncrypt utility on the unencrypted file with the "-d" option. This shows the "digest" field.

2. Look at the encrypted file using W ordPad and check the first line that shows a "Digest=…." field. If the two fields are the same then it is very likely that the encrypted and unencrypted file are the same.

Note

If a phone downloads an encrypted file that it cannot decrypt, it logs, displays an error message, and reboots. The phone will continue to do this until the boot server provides an encrypted file, an unen crypted file, or the file is removed from the master configuration file list.

For more information on this feature, refer to 3.8.4 Configuration File Encryption on page 75.

2.2.3.1 Changing the Key on the Phone

For security purposes, it may be desirable to change the key on the phones and the server from time to time.

To change a key:

1. Put the new key into a configuration file that is in the list of files downloaded by the phone (specified in 000000000000.cfg or <Ethernet address>.cfg). Use the device.sec.configEncryption.key parameter to specify the new key.

2. Manually reboot the phone so that it will download the new key. The phone will automatically reboot a second time to use the new key.

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

3. At this point the phone expects all encrypted configuration files on the boot server to use the new key and it will continue to reboot until this is the case. The files on the server must be updated to the new key or they must be made available in unencrypted format. Updating to the new key requires decrypting the file with the old key, then encrypting it with the new key. Note that configuration files, contact directory files and configuration over

ride files may all need to be updated if they were already encrypted. In the case of configuration override files, they can be deleted from the boot server so that the phone will replace them when it successfully boots.

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

Administrator’s 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 Server (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. The call duration appears in hours, minutes, and seconds.

Enable or disable all call lists or individual call lists.

• For more information, refer to 4.6.1.24 Feature <feature/ > on page 144.

None.

Administrator’s 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, if information is provided by the call server . 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 Server (if enabled)

Local Telephone User Interface

Specify whether or not to use directory name substitution.

• For more information, refer to 4.6.1.4 User Preferences <user_preferences/> on page 98.

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 through the Reset Local Config 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 that 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 Session Initiation protocol (SIP) server.

Administrator’s 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, refer to 4.6.1.24 Feature <feature/> on page 144.

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

• For more information, refer to 4.6.2.2.3 Missed Call Configuration <serverMissedCall/> on page 154.

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 Server (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, refer to 4.6.1.16 Keys <keys/> on page 132.

None.

The following diagrams and table show the default SIP key layouts for SoundPoint® IP 300, IP 301, IP 430, IP 500, IP 501, IP 600, IP 601 and SoundStation

IP 4000 models.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

SoundPoint® IP 300 and 301 Key Layout

Sel

SoundPoint® IP 430 Key Layout

28 27 25

2635

ABC21DEF

192021

DEF

ABC

MNOJKLGHI

16 17 18

GHI

PQRS

28 27

JKL

MNO

TUV87WXYZ

TUV

OPER

21 20 19

16 17

131415

WXYZ

121110

Del

Do Not Disturb

Redial

Hold

Key ID

15 14 13

10 11 12

Key ID

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

SoundPoint® IP 500 and 501 Key Layout

Conference

Directories

Services

Call Lists

Conference

Transfer

Redial

19 20 21

GHI

PQRS

SoundPoint® IP 600 and 601 Key Layout

Directories

Services

Conference

Transfer

Redial

Hold

19 20 21

GHI

PQRS

ABC21DEF

ABC

JKL

TUV87WXYZ

TUV

OPER

ABC21DEF

ABC

JKL

TUV87WXYZ

TUV

OPER

Sel

222324

DEF

MNOJKLGHI

MNO

WXYZ

Key ID

25262728

222324

DEF

MNOJKLGHI

MNO

WXYZ

131415

Do Not Disturb

Del

Messages

Hold

Sel

Del

Messages

Do Not Disturb

Hold

5 3

Key ID

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

SoundPoint® IP 4000 Key Layout

IP 600 & 601 Function

IP 4000 Function

Key ID IP 300 & 301

Function

27 28 29

1 2 3

7 8 9

13 14 15

19 20 21

IP 430 Function IP 500 & 501

Function

1 Line1 Line1 ArrowUp ArrowUp Dialpad1 2 Line2 Line2 ArrowLeft ArrowLeft Dialpad2 3 n/a n/a Select ArrowDown Dialpad3 4 n/a ArrowUp ArrowRight ArrowRight VolUp 5 Hold Hold ArrowDown Select Handsfree 6 n/a n/a Delete Delete ArrowUp 7 Redial Redial Menu Menu Dialpad4 8 VolUp VolUp Messages Messages Dialpad5 9 VolDown VolDown DoNotDisturb DoNotDisturb Dialpad6 10 DialpadStar DialpadStar Hold MicMute VolDown 11 Dialpad0 Dialpad0 VolUp VolUp n/a 12 DialpadPound DialpadPound VolDown VolDown Select 13 Dialpad9 Dialpad9 DialpadPound DialpadPound Dialpad7 14 Dialpad8 Dialpad8 Dialpad0 Dialpad0 Dialpad8 15 Dialpad7 Dialpad7 DialpadStar DialpadStar Dialpad9 16 Dialpad4 Dialpad4 Dialpad9 Dialpad9 MicMute 17 Dialpad5 Dialpad5 Dialpad8 Dialpad8 n/a 18 Dialpad6 Dialpad6 Dialpad7 Dialpad7 ArrowDown 19 Dialpad3 Dialpad3 Dialpad4 Dialpad4 DialpadStar 20 Dialpad2 Dialpad2 Dialpad5 Dialpad5 Dialpad0

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Key ID IP 300 & 301

Function

21 Dialpad1 Dialpad1 Dialpad6 Dialpad6 DialpadPound 22 n/a ArrowRight Dialpad3 Dialpad3 Redial 23 Do Not Disturb Messages Dialpad2 Dialpad2 n/a 24 n/a n/a Dialpad1 Dialpad1 n/a 25 SoftKey3 SoftKey4 SoftKey4 SoftKey4 Menu 26 MicMute Headset SoftKey3 SoftKey3 Exit 27 SoftKey2 SoftKey2 SoftKey2 SoftKey2 SoftKey1 28 SoftKey1 SoftKey1 SoftKey1 SoftKey1 SoftKey2 29 ArrowDown ArrowDown Conference Services SoftKey3 30 n/a Select CallHistory Directories n/a 31 ArrowUp ArrowLeft S ervices Line6 n/a 32 Menu n/a Directories Conference n/a 33 n/a MicMute Line3 Line2 n/a

IP 430 Function IP 500 & 501

Function

IP 600 & 601 Function

IP 4000 Function

34 n/a SoftKey3 Line2 Line1 n/a 35 Headset Handsfree Line1 Line3 n/a 36 n/a n/a Redial Redial n/a 37 n/a n/a Transfer Transfer n/a 38 n/a n/a Headset Headset n/a 39 n/a n/a MicMute Handsfree n/a 40 n/a n/a Handsfree Hold n/a 41 n/a n/a n/a Line4 n/a 42 n/a n/a n/a Line5 n/a

3.1.8 Connected Party Identification

The identity of the remote party to which the user has connected is displayed and logged, if the name and ID is provided by the call server . 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.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

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

IP phones, if using the default configuration parameters, the receive handset/headset volume resets to nominal after each call to comply with regulatory requirements. Refer to 4.6.1.8.2 Volume Persistence <volume/> on page 110.

3.1.10 Customizable Audio Sound Effects

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 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 L16/160001 (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

Central (boot server)

Web Server (if enabled)

Local

Local Phone User Interface

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

For more information, refer to:

• 4.6.1.6 Sampled Audio for Sound Effects <sampled_audio/> on page 101,

• 4.6.1.7 Sound Effects <sound_effects/> on page 103.

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 through the Reset Local Config menu selection.

None.

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

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

3.1.11 Message Waiting Indication

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

3.1.12 Distinctive Incoming Call Treatment

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

Administration: Distinctive Incoming Call Treatment

For more information, refer to 3.1.17 Local Contact Directory on page 38.

3.1.13 Distinctive Ringing

There are three options for distinctive ringing:

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

2. The ring type for specific callers can be assigned in the contact directory. For more information, refer to on page 35. This option has higher priority than option 1.

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

Configuration file: sip.cfg

Central (boot server)

Configuration file: phone1.cfg

XML File: <Ethernet address>-directory.xml

3.1.12 Distinctive Incoming Call Treatment

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

• For more information, refer to 4.6.1.1.4.2 Alert Information <alertInfo/> on page 91.

Specify the ring type to be used for each line.

• For more information, refer to 4.6.2.1 Registration <reg/> on page 149.

This file can be created manually using an XML editor.

• For more information, refer to 3.1.17.1 Local Contact Directory File Format on page 39.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Web Server (if enabled)

Local Phone User Interface

Local

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 through the Reset Local Config 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.

Central (boot server)

Configuration file: sip.cfg

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

• For more information, refer to 4.6.1.1.4.2 Alert Information <alertInfo/> on page 91.

Web Server (if enabled)

Local

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. For more information on forwarding calls while DND is enabled, refer to 3.2.5 Call Diversion (Call Forward) on page 54.

Configuration file: sip.cfg

Central (boot server)

Configuration file: phone1.cfg

None.

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

• For more information, refer to 4.6.1.12 Call Handling Configuration <call/> on page 125.

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

• For more information, refer to 4.6.2.2.1 Do Not Disturb <donotdisturb/> on page 153.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Local

Web Server (if enabled)

Local Phone User Interface

None.

Enable or disable DND using the “Do Not Disturb” key on the SoundPoint Features menu on the SoundStation

IP 300, 301, 500, 501 and 600 or the

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 430, 500, 501, 600 and 601 phones are full-duplex speakerphones. The SoundPoint® IP 300 and 301 phones are a listen-only speakerphone. The SoundPoint® phones provide dedicated keys for convenient selection of either the speakerphone or headset. The SoundStation® IP 4000

phones are full-duplex speakerphones.

Central (boot server)

Configuration file: sip.cfg

Enable or disable persistent headset mode.

• For more information, refer to 4.6.1.4 User Preferences <user_preferences/> on page 98.

IP 4000.

Local

Web Server (if enabled)

Local Phone User Interface

Enable or disable persistent headset mode. Navigate to: http://<phoneIPAddress>/coreConf.htm#us

Enable or disable persistent headset mode through 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 through the Reset Local Config menu.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

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.

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, refer to 4.6.1.13 Directory <directory/> on page 128.

A sample file named 000000000000-directory~.xml (Note the extra “~” in the filename) 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, then 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. These files can be downloaded once per reflash.

• For information on file format, refer to 3.1.17.1 Local Contact Directory File Format on page 39.

This file can be created manually using an XML editor.

• For information on file format, refer to 3.1.17.1 Local Contact Directory File Format on page 39.

Web Server (if enabled)

Local Phone User

Local

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’s 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" standa lone="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’s 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 9999 speed-dial index

rt Null, 1 to 21 ring type

dc UTF-8 encoded string cont ain-

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 the

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’s Guide - SoundPoint® IP / SoundStation® IP Features

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, refer to 4.6.1.2 Dial Plan <dialplan/> on page 94.

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, refer to 4.6.2.4 Dial Plan <dialplan/> on page 156.

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

Navigate to: http://<phoneIPAddress>/appConf.htm#ls 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 through the Reset Local Config menu selection.

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, refer to 4.6.2.1 Registration <reg/> on page 149.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Web Server (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 through the Reset Local Config 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 (refer to 3.1.20 Multiple Line Keys per Registration on page 41).

Central (boot server)

Configuration file: sip.cfg

Configuration file: phone1.cfg

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

• For more information, refer to 4.6.1.12 Call Handling Configuration <call/> on page 125.

Specify per-registration the number of calls that 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, refer to 4.6.2.1 Registration <reg/> on page 149.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Web Server (if enabled)

Local

Local Phone User Interface

Specify the default number of calls that can be active or on hold per line key and the number of calls per registration that 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

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 through the Reset Local Config menu selection.

Specify per-registration the number of calls that 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 that 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 that binds the appearances together logically and looks after the necessary state notifications and performs an access control function. For more information, refer to 5.2.4 Shared Call Appearance Signaling on page 171.

Important

Emergency routing is not supported on shared lines (refer to 4.6.1.2.2.2 Emergency <emergency/> on page 95).

Administrator’s 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, refer to 4.6.1.12.1 Shared Calls <shared/> on page 126.

Specify line-seize subscription period.

• For more information, refer to 4.6.1.1.2 Server <server/> on page 85.

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

• For more information, refer to 4.6.1.1.4.4 Special Events <specialEvent/> on page 93.

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, refer to 4.6.2.1 Registration <reg/> on page 149.

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

• For more information, refer to 4.6.2.3 Diversion <divert/> on page 154.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Local

Web Server (if enabled)

Local Phone User Interface

Specify line-seize subscription period. Navigate to: http://<phoneIPAddress>/appConf.htm#se

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

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

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

http://<phoneIPAddress>/reg.htm Changes are saved to local flash and backed up to <Ether-

net address>-phone.cfg on the boot server. They will permanently override global settings unless deleted through the Reset Local Config menu selection.

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

3.1.23 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 that 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 that binds the appearances together logically and looks after the necessary state notifications and performs an access control func-

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

tion. For more information, refer to 5.2.5 Bridged Line Appearance Signaling on page 172.

Important

Emergency routing is not supported on shared lines (refer to 4.6.1.2.2.2 Emergency <emergency/> on page 95).

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 Server (if enabled)

Specify whether diversion should be disabled on shared lines.

• For more information, refer to 4.6.1.12 Call Handling Configuration <call/> on page 125.

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, refer to 4.6.2.1 Registration <reg/> on page 149.

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

• For more information, refer to 4.6.2.3 Diversion <divert/> on page 154.

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 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 through the Reset Local Confide menu selection.

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.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

3.1.24 Busy Lamp Field

This feature is available on SoundPoint® IP 600 and 601 phones (with an attached Expansion Module) only.

The Busy Lamp Field (BLF) feature enhances support for a phone-based attendant console. It allows monitoring the hook status and remote party information of users through the busy lamp fields and displays on an attendant console phone.

Important

Do not use this feature with Microsoft® Office Live Communications Server 2005 feature (refer to

3.6.6 Microsoft® Office Live Communications Server 2005 Integration on page 69).

Important

Use this feature with TCPpreferred transport (refer to 4.6.1.1.2 Server <server/> on page 85 and

4.6.1.1.4.1 Outbound Proxy <outboundProxy/> on page 90).

Central (boot server)

Local

Configuration file: sip.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

Local Phone User Interface

None.

Specify the list SIP URI and index of the registration which will be used to send a SUBSCRIBE to the list SIP URI specified in attendant.uri.

• For more information, refer to 4.6.2.7 Attendant <attendant/> on page 161.

None.

3.1.25 Customizable Fonts and Indicators

The phone’ s user interface can be cus tomized 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

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

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

Configuration File: sip.cfg

Central (boot server)

Web Server (if enabled)

Local

Local Phone User Interface

Specify fonts to overwrite existing ones or specify new fonts.

• For more information, refer to 4.6.1.15 Fonts <font/> on page 129.

Specify which bitmaps to use.

• For more information, refer to 4.6.1.17 Bitmaps <bitmaps/ > on page 133.

Specify how to create animations and LED indicator patterns.

• For more information, refer to 4.6.1.18 Indicators <indicators/> on page 134.

None.

3.1.26 Soft Key-Driven User Interface

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

3.1.27 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

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

shown on the idle display if the SIP server supports this feature. Refer to 3.4.1 Presence on page 60.

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, refer to 3.1.17.1 Local Contact Directory File Format on page 39.

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.

3.1.28 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 Simple Network Time Protocol (SNTP) timeserver. The time and date displayed on the phone will flash continuously

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

until a successful SNTP response is 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 Server (if enabled)

Local Phone User Interface

Turn time and date display on or off.

• For more information, refer to 4.6.1.4 User Preferences <user_preferences/> on page 98.

Set the time and date display formats.

• For more information, refer to 4.6.1.3.2 Date and Time <datetime/> on page 97.

Set the basic SNTP settings and daylight savings parameters.

• For more information, refer to 4.6.1.10.2 Time Synchronization <SNTP/> on page 121.

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 through the Reset Local Config menu selection.

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

• For more information, refer to 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.

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 through the Reset Local Config menu selection.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

3.1.29 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.

Central (boot server)

Local

Configuration file: sip.cfg

Web Server (if enabled)

Local Phone User Interface

To turn idle display animation on or off.

• For more information, refer to 4.6.1.18 Indicators <indicators/> on page 134.

To replace the animation used for the idle display.

• For more information, refer to 4.6.1.18.1 Animations <Animations/> <IP_300/>, <IP_400/>, <IP_500/>, <IP_600/> and <IP_4000/> on page 134.

To change the position of the idle display animation.

• For more information, refer to 4.6.1.18.4.2 Graphic Icons <gi/> <IP_300/>, <IP_400/>, <IP_500/>, <IP_600/> and <IP_4000/> on page 136.

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)

Configuration file: phone1.cfg

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

• For more information, refer to 4.6.2.2.2 Automatic Off-hook Call Placement <autoOffHook/> on page 153.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Local

3.2.2 Call Hold

Call hold is a fundamental feature of the phone. The purpose of hold is to pause activity on one call so that the user may use the phone for another task, such as to make or receive another call. Network 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)

Web Server (if enabled)

Local Phone User Interface

Configuration file: sip.cfg

None.

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, refer to 4.6.1.1.4 SIP <SIP/> on page 88.

Specify local hold reminder options.

Web Server (if enabled)

Local

Local Phone User Interface

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.

• For more information, refer to 4.6.1.12.2 Hold, Local Reminder <hold/><localReminder/> on page 127.

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 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 through the Reset Local Config menu selection.

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).

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

• Consultation transfers that 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 or hangs up.

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, refer to 4.6.1.1.4 SIP <SIP/> on page 88.

None.

3.2.4 Three-Way Conference, Local or Centralized

Local or centralized conferences2 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 relies on network signaling.

Central (boot server)

Local

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

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, refer to 4.6.1.1.4.5 Conference Setup <conference/> on page 93.

None.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

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, calls from a specific caller (extension), when the phone is busy, when Do Not Disturb is active, 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 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 Server (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, refer to 4.6.2.3 Diversion <divert/> on page 154.

Set all call diversion settings. Navigate to: http://<phoneIPAddress>/reg.htm 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 through the Reset Local Config menu selection.

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)

Configuration file: sip.cfg

Turn this feature on or off.

• For more information, refer to 4.6.1.24 Feature <feature/> on page 144.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Web Server (if enabled)

Local

Local Phone User Interface

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 Server (if enabled)

Local Phone User Interface

None.

Turn this feature on or off.

• For more information, refer to 4.6.1.24 Feature <feature/> on page 144.

None.

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 Server (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, refer to 4.6.1.24 Feature <feature/> on page 144.

None.

Administrator’s 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 Server (if enabled)

Local

Local Phone User Interface

Turn this feature on or off.

• For more information, refer to 4.6.1.24 Feature <feature/> on page 144.

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

• For more information, refer to 4.6.1.12 Call Handling Configuration <call/> on page 125.

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 config-

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

urable for different network environments. When packets are lost, a concealment algorithm minimizes the resulting negative audio consequences.

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, refer to 4.6.1.8.1.2 Codec Profiles <profiles/> on page 109.

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

Navigate to: http://<phoneIPAddress>/coreConf.htm#au 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 through the Reset Local Config menu selection.

None.

3.3.3 Voice Activity Detection

The purpose of voice activity detection (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)

Local

Configuration file: sip.cfg

Web Server (if enabled)

Local Phone User Interface

Enable or disable VAD and set the detection threshold.

• For more information, refer to 4.6.1.8.10 V oice Activity Detection <VAD /> on page 118.

None.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

3.3.4 DTMF Tone Generation

The phone generates dual tone multi-frequency (DTMF) tones in response to user dialing on the dial pad. These tones are transmitted in the real-time transport protocol (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 end point.

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, refer to 4.6.1.5.1 Dual Tone Multi-Frequency <DTMF/> on page 99.

None.

3.3.5 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)

Configuration file: sip.cfg

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

• For more information, refer to 4.6.1.5.1 Dual Tone Multi-Frequency <DTMF/> on page 99.

Web Server (if enabled)

Local

Local Phone User Interface

None.

3.3.6 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.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

3.3.7 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 Server (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, refer to:

• 4.6.1.8.1.1 Codec Preferences <preferences/> on page 108, and

• 4.6.1.8.1.2 Codec Profiles <profiles/> on page 109.

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

Local

Local Phone User Interface

Navigate to: http://<phoneIPAddress>/coreConf.htm#au 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 through the Reset Local Config menu selection.

None.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

3.3.8 Background Noise Suppression (BNS)

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

3.3.9 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.

3.3.10 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.3 This increases the effective user-phone

radius and helps with the intelligibility of soft-talkers.

3.4 Presence and Instant Messaging Features

The phone contains both Presence and Instant Messaging features. These features are compatible with Microsoft® Windows® Messenger 5.1. 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. Users can block others from monitoring their phones and are

notified when a change in monitored status occurs4. Phone status changes are broadcast automatically to monitoring phones when the user engages in calls or invokes do-

3. AGC support will be available in a subsequent rel ease.

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

Administrator’s 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, refer to 3.1.17.1 Local Contact Directory File Format on page 39.

None.

The user can edit the directory contents. The Watch 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 render

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

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’s 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, refer to:

• 4.6.1.3.1 Multilingual <multilingual/> on page 96, and

• 4.6.1.3.1.1 Adding New Languages on page 97.

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 through the Reset Local Config menu selection.

3.5.2 Downloadable Fonts

New fonts can be loaded onto the phone. For more information, refer to 4.6.1.15 Fonts <font/> on page 129.

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’s 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, refer to 4.6.1.5.2 Chord Sets <chord_sets/> on page 100.

Specify downloaded sampled audio files for advanced call progress tones.

• For more information, refer to 4.6.1.6 Sampled Audio for Sound Effects <sampled_audio/> on page 101.

Specify patterns. For more information, refer to:

• 4.6.1.7.1 Patterns <patterns/> on page 103, and

• 4.6.1.7.1.1 Call Progress Patterns on page 105.

None.

3.6 Advanced Server Features

3.6.1 Voice Mail Integration

The phone is compatible with voice mail 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. Voice mail access can be configured to be through a single key

press (for example, the Messages key on the SoundPoint

IP 300, 301, 430, 500, 501,

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

600 and 601). A message-waiting signal from a voice mail server will trigger the message-waiting indicator to flash.

Central (boot server)

Configuration file: sip.cfg

Configuration file: phone1.cfg

Web Server (if enabled)

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

• For more information, refer to 4.6.1.4 User Preferences <user_preferences/> on page 98.

For one-touch voice mail access, choose to bypass instant messages to remove the step of selecting between instant messages and voice mail after pressing the Messages key on

the SoundPoint

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

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 voice mail.

• For more information, refer to 4. 6.2.5 Mess aging <msg / > on page 159.

For one-touch voice mail access, enable the “one-touch voice mail” user preference and choose to bypass instant messages to remove the step of selecting between instant messages and voice mail after pressing the Messages key on

the SoundPoint

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

sages are still accessible from the Main Menu).

Local

Local Phone User Interface

Navigate to: http://<phoneIPAddress>/coreConf.htm#us On a per-registration basis, specify a subscribe contact for

solicited NOTIFY applications, a callback mode (self callback or another contact) to call when the user accesses voice mail.

Navigate to: http://<phoneIPAddress>/reg.htm 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 through the Reset Local Config menu selection.

None.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

3.6.2 Multiple Registrations

SoundPoint® IP phones support multiple registrations per phone and the SoundStation® IP 4000 supports a single registration. The SoundPoint® IP 300 and 301 support a maximum of two registrations, the SoundPoint® IP 430 supports two, the SoundPoint® IP 500 and 501 support three, the SoundPoint® IP 600 supports six, and the SoundPoint® IP 601 supports 12. Up to three SoundPoint® IP Expansion Modules can

be added to a single host phone increasing the total number of buttons to 48 registrations.

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, refer to 4.6.1.1.1 Local <local/> on page 85 and 4.6.1.1.2 Server <server/> on page 85.

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, refer to 4.6.2. 1 Registratio n <reg/ > on page 149.

Administrator’s 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 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 through the Reset Local Config menu selection.

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 through the Reset Local Config menu selection.

• For more information on the fields in this menu, refer to

4.6.1.1.1 Local <local/> on page 85, 4.6.1.1.2 Server <server/> on page 85 and 4.6.2.1 Registration <reg/> on page 149.

Administrator’s 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 Server (if enabled)

Local Phone User Interface

Turn this feature on or off.

• For more information, refer to 4.6.1.24 Feature <feature/> on page 144.

Enable this feature per registration.

• For more information, refer to 4.6.2.1 Registration <reg/> on page 149.

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, refer to 4.6.1.24 Feature <feature/> on page 144.

Enable this feature per registration.

• For more information, refer to 4.6.2.1 Registration <reg/> on page 149.

Web Server (if enabled)

Local

Local Phone User Interface

None.

3.6.5 Server Redundancy

The phone can be configured with multiple SIP servers, one primary and one or more backup. The phone will switch to a backup server when the current primary server fails. Backup server configuration can be static or can use advanced DNS methods. In

Administrator’s 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 through all servers in the list and this is the last server then the signaling fails after the complete UDP timeout defined in RFC 3261. If it is not the last server in the list, the maximum number of retries using the configurable retry timeout is used. For more information, refer

4.6.1.1.2 Server <server/> on page 85 and 4.6.2.1 Registration <reg/> on

to page 149.

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.

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

Note

Failure to resolve a DNS name is treated as signalling failure that will cause a fail over.

3.6.6 Microsoft® Office Live Communications Server

SoundPoint® IP phones can used with Microsoft® Office Live Communications Server 2005 and Microsoft

cies and increase productivity and to share ideas and information immediately with business contacts.

2005 Integration

Office Communicator to help improve business efficien-

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Note

Any contacts added through the SoundPoint® IP phone’s buddy list will appear in as a contact in

Microsoft

Office Communicator and Windows® Messenger.

Important

Do not use this feature with Busy Lamp Field feature (refer to 3.1.24 Busy Lamp Field on page 47).

Central (boot server)

Configuration file: sip.cfg

Configuration file: phone1.cfg

Specify that support for Microsoft

Office Live Communi-

cations Server 2005 is enabled.

• For more information, refer to 4.6.1.1.4 SIP <SIP/> on page 88.

Specify the line/registration number used to send SUBSCRIBE for presence.

• For more information, refer to 4.6.1.14 Presence <presence/> on page 129.

Turn the presence and messaging features on or off.

• For more information, refer to 4.6.1.24 Feature <feature/> on page 144.

Specify the number of line keys to assign per registration.

• For more information, refer to 4.6.2. 1 Registratio n <reg/ > on page 149

Specify the line/registration number which has roaming buddies support enabled.

• For more information, refer to 4.6.2.8 Roaming Buddies <roaming_buddies/> on page 161.

Specify the line/registration number which has roaming privacy support enabled.

• For more information, refer to 4.6.2.9 Roaming Privacy <roaming_privacy/> on page 162.

Web Server

None.

(if enabled)

Local

Local Phone User

None.

Interface

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

3.6.6.1 Configuration File Changes

SoundPoint® IP phones can be deployed in two basic methods. In the first method, Microsoft® Office Live Communications Server 2005 serves as the call server and the

phones have a single registration. In the second method, the phone has a primary registration to call server—that is not Live Communications Server (LCS)—and a secondary registration to LCS for presence purposes.

Single Registration with Microsoft

Office Live Communications Server 2005 as

the Call Server

Modify the sip.cfg configuration file as follows:

1. Open sip.cfg in an XML editor.

2. Locate the feature parameter.

3. For the feature.1.name = presence attribute, set feature.1.enabled to 1.

4. For the feature.2.name = messaging attribute, set feature.2.enabled to 1.

5. Locate the voIpProt parameter.

6. Set the voIpProt.server.x.transport attribute to TCPpreferred or TLS. (Your selection depends on the LCS configuration.)

7. Set the voIpProt.server.x.address to the LCS address. For example, voIpProt.server.1.address = "lcs2005.local"

8. Set the voIpProt.SIP.lcs attribute to 1.

9. (Optional) If SIP forking is desired, set voIpProt.SIP.ms-forking attribute to 1. Refer to 4.6.1.1.4 SIP <SIP/> on page 88.

10. Save the modified SIP Configuration file.

Note

The TLS protocol is not supported on SoundPoint® IP 300 and 500 phones.

Modify the phone1.cfg configuration file as follows:

1. Open phone1.cfg in an XML editor.

2. Locate the registration parameter.

3. Set the reg.1.address to the LCS address. For example, reg.1.address = "7778"

4. Set the reg.1.server.y.address to the LCS server name.

5. (Optional) Set the reg.1.server.y.transport attribute to TCPpreferred or TLS. (Your selection depends on the LCS configuration.)

6. Set reg.1.auth.userId to the phone's LCS username. For example, reg.1.auth.userId = "jbloggs"

7. Set reg.1.auth.password to the LCS password. For example, reg.1.auth.password = "Password2"

8. Locate the roaming_buddies attribute.

9. Set the roaming_buddies.reg element to 1. Refer to 4.6.2.8 Roaming Buddies <roaming_buddies/> o n page 161.

10. Locate the roaming_privacy attribute.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

11. Set the roaming_privacy.reg element to 1. Refer to 4.6.2.9 Roaming Privacy <roaming_privacy/> on page 162.

12. Save the modified Per-Phone Configuration file.

Dual Registration with Microsoft® Office Live Communications Server 2005 as the Presence Server

(Optional) Modify the sip.cfg configuration file as follows:

1. Open sip.cfg in an XML editor.

2. Locate the feature parameter.

3. For the feature.1.name = presence attribute, set feature.1.enabled to 1.

4. For the feature.2.name = messaging attribute, set feature.2.enabled to 1.

5. Locate the voIpProt parameter.

6. If SIP forking is desired, set voIpProt.SIP.ms-forking attribute to 1. Refer to 4.6.1.1.4 SIP <SIP/> on page 88.

7. Save the modified SIP Configuration file.

Modify the phone1.cfg configuration file as follows:

1. Open phone1.cfg in an XML editor.

2. Locate the registration parameter.

3. Select a registration to be used for the Microsoft® Office Live Communications Server 2005. Typically, this would be 2.

4. Set the reg.x.address to the LCS address. For example, reg.2.address = "7778"

5. Set the reg.x.server.y.address to the LCS server name.

6. (Optional) Set the reg.2.server.y.transport attribute to TCPpreferred or TLS. (Your selection depends on the LCS configuration.)

7. Set reg.x.auth.userId to the phone's LCS username. For example, reg.2.auth.userId = "jbloggs"

8. Set reg.x.auth.password to the LCS password. For example, reg.2.auth.password = "Password2"

9. Locate the roaming_buddies attribute.

10. Set the roaming_buddies.reg element to the number coresponding to the LCS registration. For example, roaming_buddies.reg = 2. Refer to 4.6.2.8 Roaming Buddies <roaming_buddies/> o n page 161.

11. Locate the roaming_privacy attribute.

12. Set the roaming_privacy.reg element to the number coresponding to the LCS registration. For example, roaming_privacy.reg = 2. Refer to 4.6.2.9 Roaming Privacy <roaming_privacy/> on page 162.

13. Save the modified Per-Phone Configuration file.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

3.7 Accessory Internet Features

3.7.1 MicroBrowser

The SoundPoint® IP 600 and 601 phones support an XHTML microbrowser. This can be launched by pressing the Services key.

Central (boot server)

Local

Configuration file: sip.cfg

Web Server (if enabled)

Local Phone User Interface

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

• For more information, refer to 4.6.1.26 MicroBrowser <microbrowser/> on page 146.

Specify the Services browser home page and proxy to use. Navigate to: http://<phoneIPAddress>/coreConf.htm#mb 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 through the Reset Local Config menu selection.

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)

Configuration file: sip.cfg

Specify the minimum lengths for the user and administrator passwords.

• For more information, refer to 4.6.1.20.2 Password Lengths <pwd/><length/> on page 141.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Web Server (if enabled)

Local Phone User Interface

Local

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. Refer to

6.1 Trusted Certificate Authority List on page 173. In addition, custom certificates can be added to the phone. This is done by using the SSL Security menu on the phone to provide the URL of the custom certificate then select an option to use this custom certificate.

None.

The user and administrator passwords can be changed under the Settings menu or through configuration parameters (see

2.2.2.1.1.3 Setting Flash Parameters from Configuration

Files on page 16). 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.

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, refer to 4.6.1.1.4.3 Request Validation <requestValidation/> on page 92.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Web Server (if enabled)

Local

Local Phone User Interface

None.

3.8.4 Configuration File Encryption

Confidential information stored in configuration files must be protected from attack or unintentional discovery. This information could include registration passwords and contact information. A separate SDK is provided to facilitate key generation and configuration file encryption and decryption on a UNIX or Linux server.

The phone can recognize encrypted files, which it downloads from the boot server and it can encrypt files before uploading them to the boot server. To do this, a key must be stored on the phone. Configuration files (excluding the master configuration file), contact directories, and configuration override files can all be encrypted. The phone will still recognize unencrypted files and a combination of encrypted and unencrypted files can be used on one phone.

If the phone doesn't have a key, it must be downloaded to the phone in plain text (a potential security hole if not using HTTPS). If the phone already has a key, a new key can be downloaded to the phone encrypted using the old key (refer to 2.2.3.1 Changing the Key on the Phone on page 24). At a later date, new phones from the factory will have a key pre-loaded in them that will be shared with trusted customers. This key will be changed at regular intervals to enhance security.

Central (boot server)

Local

Configuration File: sip.cfg

Configuration file: <device>.cfg

Web Server (if enabled)

Local Phone User Interface

Specify the phone-specific contact directory and the phonespecific configuration override file.

• For more information, refer to section 4.6.1.20.1 Encryption <encryption/> on page 141.

Change the encryption key.

• For more information, refer to section 2.2.2.1.1.3 Setting Flash Parameters from Configuration Files on page 16.

None.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Note

The SoundPoint IP® 300 and 500 phones will always fail at decrypting files. These phones will recognize that a file is encrypted, but cannot decrypt it and will display an error. Encrypted configuration

files can only be decrypted on the SoundPoint IP

4000 phones.

The master configuration file cannot be encrypted on the boot server. This file is downloaded by the bootROM that does not recognize encrypted files. For more information, refer to 2.2.2.1.1.1 Master Configuration Files on page 13.

301, 430, 501, 600, and 601 and the SoundStation

Administrator’s 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® IP can be powered through a local AC power adapter or can be

line-powered (power supplied through 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® IP PC Ethernet port.

IP switch gives higher transmit priority to packets originating in the

SoundPoint® IP Switch - Port Priorities

To help ensure good voice quality, the Ethernet switch embedded in the SoundPoint® IP phones should be configured to give voice traffic emanating from the

phone higher transmit priority than those from a device connected to the PC port. If not using a VLAN (VLAN blank in the setup menu), this will automatically be the case. If using a VLAN, ensure that the 802.1p priorities for both default and real-time transport protocol (RTP) packet types are set to 2 or greater. Otherwise, these packets will compete equally with those from the PC port. For more information, refer to

4.6.1.9 Quality of Service <QOS/> on page 118.

4.2 Application Network Setup

4.2.1 Real-Time Transport Protocol 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 end points will negotiate the respective destination IP addresses and ports. This allows real-time transport control protocol (RTCP) to operate correctly even with RTP media flowing in only a single direction, or not at all. It also allows greater security: packets from unauthorized sources can be rejected.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Optimization

The phone can filter incoming R TP 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 R TP 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 R TP 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 Server (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 RT P port range start.

• For more information, refer to 4.6.1.10.3.1 RTP <RTP/ > on page 124.

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 RT P port range start.

Navigate to: http://<phoneIPAddress>/netConf.htm#rt 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 through the Reset Local Config menu selection.

None.

4.2.2 Working with Network Address Translation

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’s Guide - SoundPoint® IP / SoundStation® IP Optimization

Central (boot server)

Local

Configuration file: phone1.cfg

Web Server (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, refer to 4.6.2.6 Network Address Translation <nat/> on page 160.

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 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 through the Reset Local Config menu selection.

None.

4.3 Updating and Rebooting

The bootROM, application executable, and configuration files can be updated automatically through the centralized provisioning (boot server) model. There files are read-only by default.

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 that 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 configu ration files.

Important

The configuration files listed in CONFIG_FILES attribute of the master configuration file must be updated when the software is updated. Any new configuration files must be added to the COBFIG_FILES attribute in the appropriate order.

For more information, refer to the “Configuration File Management on SoundPoint® IP Phones” whitepaper at www.polycom.com/support/voip/ .

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

4. Reboot the phones. Refer to Manual Reboot: Menu Option or Key Presses on page 80.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Optimization

For more information, refer to 2.2.2 Application Configuration on page 13. For the latest Release Notes for system requirements (bootROM version for each

SoundPoint® IP and SoundStation® IP), go to www.polycom.com/support.

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. 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 430, 500, and 501:

IP 600 and 601:

IP 4000:

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

4.6.1.1.4.4 Special Events <specialEvent/> on page 93.

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 4.6.1.21 Provisioning <provisioning/> on page 142.

4.4 Event Logging

Volume-, Volume+, Hold, Do Not Disturb

Volume-, Volume+, Hold, Messages

Volume-, Volume+, Mute, Messages

*, #, Volume+, Select

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 append5 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.

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

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Optimization

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

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 502, 600, and 601:

SoundStation

IP 430, 500,

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, refer to 4.6.1.19 Event Logging <logging/> on page 137.

Specify different directory to use for log files if desired.

• For more information, refer to 2.2.2.1.1.1 Master Configuration Files on page 13.

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 be

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Optimization

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 for one

of the following reasons

• When it has a valid VLAN ID set in its network configuration

• When it is instructed to tag packets through Cisco Discovery Protocol (CDP) running on a connected Ethernet switch

• When a VLAN ID is obtained from DHCP (refer to 2.2.1.3.2 DHCP Menu on page 9)

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, refer to 4.6.1.9.2 IP TOS <IP/> on page 119.

Specify IP TOS settings. Navigate to: http://<phoneIPAddress>/netConf.htm#qo

None.

Central (boot server)

Local

Configuration file: sip.cfg

Web Server (if enabled)

Local Phone User Interface

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

• For more information, refer to 4.6.1.9.1 Ethernet IEEE

802.1p/Q <Ethernet/> on page 118.

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, refer to 2.2.1 Basic Network Setup on page 5.

Administrator’s 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’s 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

Note

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. Refer to 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 through 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.

For more information, refer to 2.2.2.1.1 Configuration Files on page 13 and 2.2.2.2 Local Phone Configuration on page 22.

Important

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.

Administrator’s 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.dhcp.available 0, 1 0 If set to 1, check with the

voIpProt.server.dhcp.option 128 to 255 Op tion to request from the

voIpProt.server.dhcp.type 0, 1 If set to 0, IP request address.

Values

Default Interpretation

DHCP server for SIP server IP address. If set to 0, do not check with DHCP server.

DHCP server if voIpProt.server.dhcp.available = 1. There is no default value for this parameter, it must be filled in with a valid value.

If set to 1, request string. Type to request from the

DHCP server if voIpProt.server.dhcp.available = 1.

There is no default value for this parameter, it must be filled in with a valid value.

Administrator’s Guide - SoundPoint IP / SoundStation® IP Optimization

Permitted

Attribute

Values

Default Interpretation

voIpProt.server.x.address dotted-decimal

IP address or host name

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

65535

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.tran sport 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’s Guide - SoundPoint® IP / SoundStation® IP Optimization

Permitted

Attribute

Values

Default Interpretation

voIpProt.server.x.transport DNSnaptr or

TCPpreferred or UDPonly or TLS

DNSna ptr

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.

If set to TLS: If TLS fails, transport fails. Leave port field empty (will default to 5061) or set to 5061. Note: TLS is not supported on

SoundPoint phones.

IP 300 and 500

voIpProt.server.x.expires positive integer,

minimum 300

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

minimum 5, maximum 65535

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

voIpProt.server.x.retryTimeOut Null or

non-negative integer

3600 Requested registration period

in seconds

60 The num ber of seconds before

the expiration time returned by server x at which the phone should try to re-register. The phone will try to re-register at half the expiration time returned by the server if that value is less than the configured overlap value.

an outbound proxy without registration.

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).

Administrator’s Guide - SoundPoint IP / SoundStation® IP Optimization

Permitted

Attribute

Values

Default Interpretation

voIpProt.server.x.retryMaxCount Null or

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

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 at the beginning of the overlap period. For example, if “expires”=3600 and “overlap”=60, the phone will re-register after 3540 seconds (3600 – 60).

4.6.1.1.3 SDP <SDP/>

Attribute

volpProt.SDP.answer.userLocalPreferences

3 If set to 0 or Null, 3 is used. non-negative integer

30 Requested line-seize subscripminimum 10

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

tion period.

Permitted Values

0 or 1 0 If set to 1, the phones uses its own pref-

Default Interpretation

erence list when deciding which codec to use rather than the preference list in the offer. If set to 0, disabled.

4.6.1.1.4 SIP <SIP/>

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”

Permitted 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 Microsoft

Office Live Communications Server 2005.

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Optimization

Permitted

Attribute

voIpProt.SIP.ms-forking 0, 1 0 If set to 0, support for MS-forking is

voIpProt.SIP.dialog.usePvalue 0, 1 0 If set to 0, phone uses "pval" field

Values

Default Interpretation

disabled. If set to 1, support for MSforking is enabled and the phone will reject all Instant Message INVITEs. This parameter is relevant for

Microsoft tions Server 2005 server installations.

Note that if any end point registered to the same account has MS-forking disabled, all other end points default back

to non-forking mode. Windows senger does not use MS-forking so be aware of this behavior if one of the end

points is Windows

name in Dialog. This obeys the draftietf-sipping-dialog-package-06.txt draft. If set to 1, phone uses a field name of "pvalue".

Office Live Communica-

Messenger.

Mes-

voIpProt.SIP.connectionReuse.useAlias

0, 1 0 If set to 0, shows old behavior.

If set to 1, phone uses the connection reuse draft which introduces "alias".

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

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

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

voIpProt.SIP.keepalive.sessionTimers

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

enabled. If set to 0, the session timer will be dis-

abled, 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 end point asks for it.

voIpProt.SIP.requestURI.E164.addGlobalPrefix

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

E.164 user parts in sip: URIs:.

Administrator’s Guide - SoundPoint IP / SoundStation® IP Optimization

Permitted

Attribute

Values

Default Interpretation

voIpProt.SIP.allowTransferOnProceeding

voIpProt.SIP.dialog.useSDP 0, 1 0 If set to 0, new dialog event package

voIpProt.SIP.pingInterval 0 to 3600 0 The number in seconds to send "PING"

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

4.6.1.1.4.1 Outbound Proxy <outboundProxy/>

Permitted

Attribute

Values

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

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

draft is used (no SDP in dialog body). If set to 1, for backwards compatibility,

use this setting to send SDP in dialog body.

message. This feature is disabled by default.

Default Interpretation

voIpProt.SIP.outboundProxy.address dotted-deci-

mal IP address or host name

voIpProt.SIP.outboundProxy.port 1 to 65535 5060

Null IP address or host name and

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

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Optimization

Permitted

Attribute

Values

Default Interpretation

voIpProt.SIP.outboundProxy.transport DNSnaptr or

TCPpreferred or UDPonly or TLS

DNSnaptrIf set to Null or DNSnaptr:

If voIpProt.SIP.outboundProxy.address is a hostname and voIpProt.SIP.outboundProxy.port is 0 or Null, do NAPTR then SRV loo k-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.

If set to TLS: If TLS fails, transport fails. Leave port field empty (will default to 5061) or set to

5061.

4.6.1.1.4.2 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

Note: TLS is not supported on SoundPoint

500 phones.

IP 300 and

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 (refer to

4.6.1.7.2 Ring type <ringType/> on page 107) will be applied.

Administrator’s Guide - SoundPoint IP / SoundStation® IP Optimization

4.6.1.1.4.3 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.

WARNING: Intensive request validation may have a negative performance impact due to the additional signaling required in some cases, therefore, use it judiciously.

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;

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

voIpProt.SIP.requestValidation.digest.realm

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

both or all: apply both of the above methods

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.

A valid string PolycomSPIP Determines string used for

Realm.

Polycom 1725-11530-200 User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

Table of Contents

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

2.2.1.4 Reset to Factory Defaults

2.2.2 Application Configuration

2.2.2.1 Centralized Configuration

2.2.2.2 Local Phone Configuration

2.2.3 Management of File Encryption and Decryption

2.2.3.1 Changing the Key on the Phone

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 Busy Lamp Field

3.1.25 Customizable Fonts and Indicators

3.1.26 Soft Key-Driven User Interface

3.1.27 Speed Dial

3.1.28 Time and Date Display

3.1.29 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 Voice Activity Detection

3.3.4 DTMF Tone Generation

3.3.5 DTMF Event RTP Payload

3.3.6 Acoustic Echo Cancellation (AEC)

3.3.7 Audio Codecs

3.3.8 Background Noise Suppression (BNS)

3.3.9 Comfort Noise Fill

3.3.10 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

3.5.3 Synthesized Call Progress Tones

3.6 Advanced Server Features