Polycom Car Speaker 1725 11530 200 Rev A1 User Manual
Administrator’s Guide  
®
®
SoundPoint /SoundStation IP SIP  
Version 2.0  
August 2006  
Copyright © 2006 Polycom, Inc. All rights reserved.  
Administrator’s Guide - SoundPoint® IP / SoundStation® IP  
Table of Contents  
Table of Contents  
1 Overview......................................................... 1  
2 Installation and Operation ................................. 3  
2.1 Installation Models..................................................3  
2.2 Installation Process..................................................4  
2.2.1 Basic Network Setup....................................................................................................................... 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  
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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  
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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  
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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  
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Copyright © 2006 Polycom, Inc.  
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  
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Copyright © 2006 Polycom, Inc.  
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 separately, the behavior and configuration of the SoundPoint IP 301 is  
®
the same as the 300, the behavior and configuration of the SoundPoint IP 501 is the same as the 500,  
®
the behavior and configuration of the SoundPoint IP 601 is the same as the 600.  
SoundPoint® IP and SoundStation® IP are feature-rich, enterprise-class voice commu-  
nications terminals for Ethernet TCP/IP networks. They are designed to facilitate high-  
quality audio communications. These phones are end points in the overall network  
topology designed to interoperate with other compatible equipment including applica-  
tion servers, media servers, internetworking gateways, voice bridges, and other end  
points.  
Copyright © 2006 Polycom, Inc.  
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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 pro-  
tocols 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, confer-  
ence, forward, voice mail compatibility, and contact directory.  
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Copyright © 2006 Polycom, Inc.  
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, cen-  
trally 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 per-  
phone configuration files. To augment the central provisioning model or as the sole  
method in smaller systems, configuration can be done using user interfaces driven  
from the phones themselves: both a local setup user interface and a web server-based  
user interface are available to make configuration changes.  
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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 automati-  
cally.  
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 reposi-  
tory for:  
• boot process and application event log files - very effective when diag-  
nosing 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 infor-  
mation, refer to 2.2.1 Basic Network Setup on page 5.  
2. Application Configuration such as application specific parameters. For  
more information, refer to 2.2.2 Application Configuration on page 13.  
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.  
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Administrator’s Guide - SoundPoint® IP / SoundStation® IP  
Installation and Operation  
To 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 net-  
work 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:  
Configuration File  
Local  
Parameter  
IP address  
DHCP Optiona  
DHCP  
(Phase 2: application only)  
FLASH  
ꢀpriority when more than one source exists ꢀ  
1
•
2
-
3
•
1
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Administrator’s Guide - SoundPoint® IP / SoundStation® IP  
Installation and Operation  
Configuration File  
(Phase 2: application only)  
Local  
FLASH  
Parameter  
DHCP Optiona  
DHCP  
1
3
subnet mask  
IP gateway  
•
•
•
-
-
-
•
•
•
Refer to  
2.2.1.3.2  
boot server address  
DHCP Menu  
on page 9  
b
SIP server address  
•
•
•
-
•
•
•
151  
c
42 then 4  
2
SNTP server address  
•
d
SNTP GMT offset  
•
6
6
DNS server IP address  
•
•
-
-
•
•
alternate DNS server IP  
address  
15  
DNS domain  
•
-
•
e
Refer to  
2.2.1.3.2  
DHCP Menu  
on page 9  
Special Case: Cisco Discovery Protocol (CDP) over-  
rides Local FLASH that overrides DHCP VLAN Dis-  
covery.  
VLAN ID  
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 configura-  
tion files, uploading and downloading the configuration override file and user direc-  
tory, 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® and  
SoundStation® phones. The SoundPoint® IP 301, 430, 501, 600 and 601 and  
SoundStation® IP 4000 bootROM also supports HTTP, while the SIP application sup-  
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Copyright © 2006 Polycom, Inc.  
 
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 Type  
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  
300, 500  
301, 430, 501, 300, 500 301, 430, 501,  
600, 601,  
4000  
600, 601,  
4000  
FTP  
FTP  
FTP  
FTP  
FTP  
TFTP  
HTTP  
HTTPS  
TFTP  
FTP  
FTP  
TFTP  
HTTP  
HTTP  
TFTP  
HTTP  
TFTP  
HTTP  
Not supported. Trans- 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).  
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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  
Phase 2: application  
The network configuration menu is accessible during the  
auto-boot countdown of the bootROM phase of operation.  
Press the SETUP soft key to launch the main menu.  
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.  
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 parame-  
ters 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 sub-  
net 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 Valuesa  
Description  
DHCP Client  
Enabled, Disabled  
If enabled, DHCP will be used to obtain the  
parameters discussed in 2.2.1.1 DHCP or Man-  
ual TCP/IP Setup on page 5.  
DHCP Menu  
Refer to 2.2.1.3.2 DHCP Menu on page 9.  
Note: Disabled when DHCP client is disabled.  
Phone IP Address  
Subnet Mask  
dotted-decimal IP  
address  
Phone’s IP address.  
Note: Disabled when DHCP client is enabled.  
dotted-decimal subnet  
mask  
Phone’s subnet mask.  
Note: Disabled when DHCP client is enabled.  
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Administrator’s Guide - SoundPoint® IP / SoundStation® IP  
Installation and Operation  
Name  
Possible Valuesa  
Description  
IP Gateway  
dotted-decimal IP  
address  
Phone’s default router.  
Server Menu  
Refer to 2.2.1.3.3 Server Menu on page 11.  
SNTP Address  
dotted-decimal IP  
address  
Simple Network Time Protocol (SNTP) server  
from which the phone will obtain the current  
time.  
OR  
domain name string  
GMT Offset  
DNS Server  
-13 through +12  
Offset of the local time zone from Greenwich  
Mean Time (GMT) in half hour increments.  
dotted-decimal IP  
address  
Primary server to which the phone directs  
Domain Name System (DNS) queries.  
DNS Alternate Server dotted-decimal IP  
address  
Secondary server to which the phone directs  
Domain Name System queries.  
DNS Domain  
Ethernet  
domain name string  
Phone’s DNS domain.  
Refer to 2.2.1.3.4 Ethernet Menu on page 12.  
b
Enabled, Disabled  
This parameter is relevant if the phone gets  
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).  
EM Power  
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 configu-  
ration parameters are described in the following table:  
Possible  
Values  
Name  
Description  
Timeout  
1 through 600  
Number of seconds the phone waits for secondary  
DHCP Offer messages before selecting an offer.  
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Possible  
Values  
Name  
Description  
Boot Server  
Option 66  
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 informa-  
tion 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  
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  
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  
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.  
Boot Server Option  
128 through 254 When the boot server parameter is set to Custom, this  
(Cannot be the  
same as VLAN  
ID Option)  
parameter specifies the DHCP option number in which  
the phone will look for its boot server.  
Boot Server Option  
Type  
IP Address  
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  
String  
VLAN Discovery  
Disabled  
Fixed  
No VLAN discovery through DHCP.  
Use predefined DHCP private option values of 128,  
144, 157 and 191. If this is used, the VLAN ID Option  
field will be ignored.  
Custom  
Use the number specified in the VLAN ID Option field  
as the DHCP private option value.  
VLAN ID Option  
128 through 254 The DHCP private option value (when VLAN Discov-  
(Cannot be the  
same as Boot  
Server Option)  
ery is set to Custom).  
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2.2.1.3.3 Server Menu  
Name  
Possible Values  
Description  
Server Type  
FTP or Trivial FTP or  
HTTP or HTTPS  
The protocol that the phone will use to obtain con-  
figuration and phone application files from the  
boot server. Refer to 2.2.1.2 Provisioning File  
Transfer on page 6.  
Server Address  
dotted-decimal IP address The boot server to use if the DHCP client is dis-  
OR  
abled, 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.  
domain name string  
OR  
URL  
All addresses can be fol-  
lowed by an optional  
directory and optional file  
name.  
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 User  
Server Pass-  
any string  
any string  
The user name used when the phone logs into the  
server (if required) for the selected Server Type.  
Note: If the Server Address is a URL with a user  
name, this will be ignored.  
The password used when the phone logs in to the  
server if required for the selected Server Type.  
a
word  
Note: If the Server Address is a URL with user  
name and password, this will be ignored.  
File Transmit  
Tries  
1 to 10  
Default 3  
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.)  
Retry Wait  
0 to 300  
Default 1  
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.  
Provisioning  
Default or SAS-VP  
If SAS-VP is selected, provisioning is done (in  
addition to the normal process).  
b
Method  
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Name  
Possible Values  
Description  
Provisioning  
any string  
The URL used in XML post/response transac-  
tions. If empty, the configured URL is used.  
b
String  
This field is disabled when Provisioning Method  
is Default.  
a. The server user name and password should be changed from the default values. Note that  
for insecure protocols the user chosen should have very few privileges on the server.  
®
®
b. Not available on SoundPoint IP 300 and SoundPoint IP 500 phones.  
2.2.1.3.4 Ethernet Menu  
Name  
Possible Values  
Description  
CDP  
Enabled, Disabled  
If enabled, the phone will use CDP. It also reports  
power usage to the switch.  
VLAN ID  
Null, 0 through 4094  
Phone’s 802.1Q VLAN identifier.  
Note: Null = no VLAN tagging  
a
Auto, 10HD, 10FD,  
100HD, 100FD  
The network speed over the Ethernet.  
The default value is Auto.  
LAN  
a
Auto, 10HD, 10FD,  
100HD, 100FD  
The network speed over the Ethernet.  
The default value is Auto.  
PC  
®
a. Only available on SoundPoint IP 430 and 601 phones. HD means half duplex and FD  
means full duplex.  
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).  
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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 auto-  
matically 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 con-  
figuration 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 cus-  
tomization), 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  
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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”?>  
<!-- Default Master SIP Configuration File -->  
<!-- edit and rename this file to <Ethernet-address>.cfg for each  
phone. -->  
<!-- $Revision: 1.14 $ $Date 2005/07/27 18:43:30 $ -->  
< 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.  
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CONTACTS_DIRECTOR  
Y
An alternative directory to use for user directory files if required. A  
URl can also be specified. This is blank by default.  
OVERRIDES_DIRECTO  
RY  
An 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 IP Phones”  
whitepaper at www.polycom.com/support/voip/.  
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  
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Category  
Description  
Example  
User / per-  
phone  
Contains parameters unique to a particular phone user. Typical  
parameters include:  
phone1.cfg  
•
•
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 run-  
ning 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 spe-  
cific release of the SIP software must be used together with that software. Failure to do this may ren-  
der 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.  
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 Con-  
figuration 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.  
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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 con-  
figuration 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  
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.  
device.xxx.yyy.set  
0 or 1  
default = 0  
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.  
device.net.ipAddress  
device.net.subnetMask  
device.net.IPgateway  
dotted-decimal IP address Phone's IP address.  
Note: This field is not used when DHCP client is  
enabled.  
dotted-decimal IP address Phone's subnet mask.  
Note: This field is not used when DHCP client is  
enabled.  
dotted-decimal IP address Phone's default router / IP gateway.  
Note: This field is not used when DHCP client is  
enabled.  
device.net.vlanId  
Null, 0 to 4094  
0 or 1  
Phone’s 802.1Q VLAN identifier.  
Note: Null = no VLAN tagging  
device.net.cdpEnabled  
If set to 1, the phone will attempt to determine its  
VLAN ID through the CDP.  
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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.bootSrvU-  
seOpt  
0 to 3  
For descriptions, refer to 2.2.1.3.2 DHCP Menu  
on page 9.  
device.dhcp.bootSrvOpt  
128 to 254 (Cannot be the  
same as VLAN ID  
Option)  
device.dhcp.bootSrvOpt-  
Type  
0 or 1  
device.dhcp.dhcpVlan-  
DiscUseOpt  
0 to 2  
device.dhcp.dhcpVlan-  
DiscOpt  
128 to 254 (Cannot be the  
same as Boot Server  
Option)  
device.prov.serverName  
device.prov.serverType  
device.prov.user  
any string  
0 to 4  
For descriptions, refer to 2.2.1.3.3 Server Menu  
on page 11  
any string  
any string  
0 or 1  
device.prov.password  
device.prov.appProvType  
device.prov.app-  
ProvString  
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  
device.dns.serverAddress  
device.dns.altSrvAddress  
device.dns.domain  
-43200 to 46800  
GMT offset in seconds, corresponding to -12 to  
+13 hours.  
dotted-decimal IP address Primary server to which the phone directs Domain  
Name System queries.  
dotted-decimal IP address Secondary server to which the phone directs  
Domain Name System queries.  
any string  
any string  
The phone’s DNS domain.  
device.auth.localAdmin-  
Password  
The phone’s local administrator password.  
device.auth.localUser-  
Password  
any string  
The phone user’s local password.  
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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  
any string  
The SIP registration password for registration x  
where x = 1 to 12.  
device.sec.configEncryp-  
tion.key  
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 Sound-  
Point® 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 serv-  
ers 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 avail-  
able 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.  
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Step:  
Instructions:  
1. Set up boot server(s).  
Install boot server application or locate suitable exist-  
ing server(s). Use RFC-compliant servers.  
a
Create account and home directory. Note that each  
Note:  
phone may open multiple connections to the server.  
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.  
The phone will attempt to upload log files, a configu-  
ration 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 func-  
tion 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.  
2. Copy all files.  
Copy all files from the distribution zip file to the  
phone home directory. Maintain the same folder hier-  
archy.  
3. Create per-phone configuration files.  
Obtain a list of phone Ethernet addresses (barcoded  
label on underside of phone).  
Create per-phone phoneXXXX.cfg and <Ethernet  
address>.cfg files by using the 00000000000.cfg and  
phone1.cfg files from the distribution as templates.  
Note:  
This step may be omitted if per-phone con-  
figuration is not needed.  
Edit contents of phoneXXXX.cfg as appropriate. For  
example, edit the registration parameters.  
Edit the CONFIG_FILES attribute of the <Ethernet  
address>.cfg files so that it references the appropriate  
phoneXXXX.cfg file. (Replace the reference to  
phone1.cfg with phoneXXXX.cfg.)  
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Step:  
Instructions:  
4. Create a new configuration file (in the  
style of sip.cfg).  
For more information on why to create another config-  
uration 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 cor-  
rect 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 dic-  
tates.  
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 applica-  
tion executable and the standard configuration files,  
should be made read-only through file server file per-  
missions.  
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Step:  
Instructions:  
6. Reboot phones after configuring their  
boot server through DHCP or statically.  
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 Mes-  
sages  
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.  
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  
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 ban-  
ner.  
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.  
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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 pass-  
word is used.  
Factory default passwords are:  
•
•
User password: 123  
Administrator password: 456  
Passwords:  
Administrator password  
required.  
Network Configuration  
SIP Configuration  
SSL Security settings  
Reset to Default - local configuration, device settings, and file sys-  
tem format  
User password required.  
Restart Phone  
Changes made through the web server or local user interface are stored internally as  
overrides. These overrides take precedence over settings contained in the configura-  
tion 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 encryp-  
tion 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 con-  
figuration file encryption and decrypt on a UNIX or Linux server. The utility is distrib-  
uted 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  
Copyright © 2006 Polycom, Inc.  
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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;Key-  
Desc=companyNameKey1;Key=06a9214036b8a15b512e03d534120006;  
It is recommended that all keys have unique descriptive strings in order to allow sim-  
ple identification of which key was used to encrypt a file. This makes boot server man-  
agement 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 direc-  
tory 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 WordPad 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.  
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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.  
Copyright © 2006 Polycom, Inc.  
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Installation and Operation  
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®
Administrator’s Guide - SoundPoint IP / SoundStation IP  
Features  
3 Features  
This section describes the many features and corresponding administration points of  
SoundPoint® IP and SoundStation® IP. References are made frequently to 4.6 Config-  
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.  
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.  
Configuration File: Enable or disable all call lists or individual call lists.  
Central  
sip.cfg  
(boot  
•
For more information, refer to 4.6.1.24 Feature <feature/  
> on page 144.  
server)  
Web Server  
(if enabled)  
None.  
None.  
Local  
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.  
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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.  
Configuration File: Specify whether or not to use directory name substitution.  
Central  
sip.cfg  
(boot  
•
For more information, refer to 4.6.1.4 User Preferences  
<user_preferences/> on page 98.  
server)  
Web Server  
(if enabled)  
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 perma-  
nently override global settings unless deleted through the  
Reset Local Config menu selection.  
Local  
Local Telephone  
User Interface  
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.  
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Features  
Configuration file:  
sip.cfg  
Turn this feature on or off.  
For more information, refer to 4.6.1.24 Feature  
<feature/> on page 144.  
•
Central  
Configuration file:  
phone1.cfg  
Specify per-registration whether all missed-call events  
or only remote/server-generated missed-call events will  
be displayed.  
(boot  
server)  
•
For more information, refer to 4.6.2.2.3 Missed Call  
Configuration <serverMissedCall/> on page 154.  
Web Server  
None.  
(if enabled)  
Local  
Local Phone User  
None.  
Interface  
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.  
Configuration File:  
sip.cfg  
Set the key scrolling timeout, key functions, and sub-point-  
ers for each key (usually not necessary).  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.16 Keys <keys/> on  
page 132.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Telephone  
User Interface  
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.  
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Features  
SoundPoint® IP 300 and 301 Key Layout  
1
2
28  
27  
25  
31  
29  
32  
35  
26  
ABC  
DEF  
Menu  
1 21  
2
3 19  
20  
Do Not Disturb  
Redial  
GHI  
JKL  
MNO  
23  
7
4 16 5 17 6 18  
PQRS  
TUV  
WXYZ  
Hold  
5
7 15 8 14 9 13  
OPER  
0
10  
11  
12  
9
8
Key ID  
SoundPoint® IP 430 Key Layout  
1
2
4
25  
28  
27  
34  
22  
31 30  
29  
35  
26  
19  
33  
32  
21  
20  
17  
14  
11  
23  
7
16  
15  
10  
18  
13  
12  
5
9
8
Key ID  
30  
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®
Administrator’s Guide - SoundPoint IP / SoundStation IP  
Features  
SoundPoint® IP 500 and 501 Key Layout  
1
35  
Sel  
Del  
40  
39  
2
4
3
5
34  
33  
38 Directories
6
32  
Services  
31  
Menu  
28  
27  
26  
25  
7
8
Call Lists  
30  
ABC  
DEF  
Messages  
1
2
3
Conference  
Transfer  
Redial  
24  
23  
22  
29  
37  
36  
Do Not Disturb  
GHI  
JKL  
MNO  
9
4 19 5 20 6 21  
PQRS  
TUV  
WXYZ  
Hold  
718 8 17 9 16  
10  
OPER  
0 14  
13  
15  
11  
12  
Key ID  
SoundPoint® IP 600 and 601 Key Layout  
34  
33  
35  
41  
1
2
5
3
4
42  
31  
6
Menu  
Directories  
30  
7
28  
27  
26  
25  
Messages  
Services  
Conference  
Transfer  
Redial  
ABC  
DEF  
29  
32  
37  
36  
8
2
3
1 24  
Do Not Disturb  
23  
20  
17  
14  
22  
9
GHI  
JKL  
MNO  
5
6
4 19  
21  
16  
10  
PQRS  
TUV  
WXYZ  
7
8
9
39  
38  
18  
OPER  
Hold  
0
40  
15  
13  
11  
12  
Key ID  
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Features  
SoundPoint® IP 4000 Key Layout  
6
25  
12  
18  
26  
27  
28  
29  
4
1
2
8
3
5
10  
7
9
16  
22  
13  
19  
14  
20  
15  
21  
Key ID  
IP 300 & 301  
Function  
IP 430 Function  
IP 500 & 501  
Function  
IP 600 & 601  
Function  
IP 4000 Function  
1
Line1  
Line1  
ArrowUp  
ArrowLeft  
Select  
ArrowUp  
ArrowLeft  
ArrowDown  
ArrowRight  
Select  
Dialpad1  
Dialpad2  
Dialpad3  
VolUp  
2
Line2  
Line2  
3
n/a  
n/a  
4
n/a  
ArrowUp  
Hold  
ArrowRight  
ArrowDown  
Delete  
5
Hold  
Handsfree  
ArrowUp  
Dialpad4  
Dialpad5  
Dialpad6  
VolDown  
n/a  
6
n/a  
n/a  
Delete  
7
Redial  
Redial  
Menu  
Menu  
8
VolUp  
VolUp  
Messages  
DoNotDisturb  
Hold  
Messages  
DoNotDisturb  
MicMute  
VolUp  
9
VolDown  
DialpadStar  
Dialpad0  
DialpadPound  
Dialpad9  
Dialpad8  
Dialpad7  
Dialpad4  
Dialpad5  
Dialpad6  
Dialpad3  
Dialpad2  
VolDown  
DialpadStar  
Dialpad0  
DialpadPound  
Dialpad9  
Dialpad8  
Dialpad7  
Dialpad4  
Dialpad5  
Dialpad6  
Dialpad3  
Dialpad2  
10  
11  
12  
13  
14  
15  
16  
17  
18  
19  
20  
VolUp  
VolDown  
DialpadPound  
Dialpad0  
DialpadStar  
Dialpad9  
Dialpad8  
Dialpad7  
Dialpad4  
Dialpad5  
VolDown  
DialpadPound  
Dialpad0  
DialpadStar  
Dialpad9  
Dialpad8  
Dialpad7  
Dialpad4  
Dialpad5  
Select  
Dialpad7  
Dialpad8  
Dialpad9  
MicMute  
n/a  
ArrowDown  
DialpadStar  
Dialpad0  
32  
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Administrator’s Guide - SoundPoint IP / SoundStation IP  
Features  
Key ID  
IP 300 & 301  
Function  
IP 430 Function  
IP 500 & 501  
Function  
IP 600 & 601  
Function  
IP 4000 Function  
21  
22  
23  
24  
25  
26  
27  
28  
29  
30  
31  
32  
33  
34  
35  
36  
37  
38  
39  
40  
41  
42  
Dialpad1  
n/a  
Dialpad1  
ArrowRight  
Messages  
n/a  
Dialpad6  
Dialpad3  
Dialpad2  
Dialpad1  
SoftKey4  
SoftKey3  
SoftKey2  
SoftKey1  
Conference  
CallHistory  
Services  
Directories  
Line3  
Dialpad6  
Dialpad3  
Dialpad2  
Dialpad1  
SoftKey4  
SoftKey3  
SoftKey2  
SoftKey1  
Services  
Directories  
Line6  
DialpadPound  
Redial  
n/a  
Do Not Disturb  
n/a  
n/a  
SoftKey3  
MicMute  
SoftKey2  
SoftKey1  
ArrowDown  
n/a  
SoftKey4  
Headset  
SoftKey2  
SoftKey1  
ArrowDown  
Select  
Menu  
Exit  
SoftKey1  
SoftKey2  
SoftKey3  
n/a  
ArrowUp  
Menu  
n/a  
ArrowLeft  
n/a  
n/a  
Conference  
Line2  
n/a  
MicMute  
SoftKey3  
Handsfree  
n/a  
n/a  
n/a  
Line2  
Line1  
n/a  
Headset  
n/a  
Line1  
Line3  
n/a  
Redial  
Redial  
n/a  
n/a  
n/a  
Transfer  
Headset  
MicMute  
Handsfree  
n/a  
Transfer  
Headset  
Handsfree  
Hold  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
Line4  
n/a  
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.  
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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 nomi-  
nal 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 customi-  
zable. 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 for-  
mat. 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: Specify patterns used for sound effects and the individual  
sip.cfg  
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.  
Web Server  
(if enabled)  
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.  
Local  
Local Phone User  
Interface  
None.  
®
®
1. L16/16000 is not supported on SoundPoint IP 300, 301 and SoundStation IP 4000 phones.  
34  
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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 3.1.12 Distinctive Incoming Call Treatment  
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  
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.  
Configuration file:  
phone1.cfg  
Specify the ring type to be used for each line.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.2.1 Registration  
<reg/> on page 149.  
XML File: <Ethernet  
address>-direc-  
tory.xml  
This file can be created manually using an XML editor.  
•
For more information, refer to 3.1.17.1 Local Con-  
tact Directory File Format on page 39.  
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Features  
Web Server  
(if enabled)  
None.  
Local Phone User  
Interface  
The user can edit the ring types selected for each line  
under the Settings menu. The user can also edit the  
directory contents.  
Local  
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, cur-  
rently limited to two styles.  
Configuration file:  
sip.cfg  
Specify the mapping of Alert-Info strings to call waiting  
types.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.1.4.2 Alert  
Information <alertInfo/> on page 91.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
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  
Specify whether or not DND results in incoming calls  
being given busy treatment.  
•
For more information, refer to 4.6.1.12 Call Han-  
dling Configuration <call/> on page 125.  
Central  
(boot  
server)  
Configuration file:  
phone1.cfg  
Specify whether DND is treated as a per-registration fea-  
ture or a global feature on the phone.  
•
For more information, refer to 4.6.2.2.1 Do Not Dis-  
turb <donotdisturb/> on page 153.  
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Features  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
Enable or disable DND using the “Do Not Disturb” key  
®
on the SoundPoint IP 300, 301, 500, 501 and 600 or the  
®
Features menu on the SoundStation IP 4000.  
3.1.16 Handset, Headset, and Speakerphone  
SoundPoint® IP phones come standard with a handset and a dedicated connector is  
provided for a headset (not supplied). The SoundPoint® IP 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 conve-  
nient selection of either the speakerphone or headset. The SoundStation® IP 4000  
phones are full-duplex speakerphones.  
Configuration file:  
sip.cfg  
Enable or disable persistent headset mode.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.4 User Prefer-  
ences <user_preferences/> on page 98.  
Web Server  
(if enabled)  
Enable or disable persistent headset mode.  
Navigate to: http://<phoneIPAddress>/coreConf.htm#us  
Local Phone User  
Interface  
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.  
Local  
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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.  
Configuration file:  
sip.cfg  
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.  
XML file:  
000000000000-direc-  
tory.xml  
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>-direc-  
tory.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 con-  
tents, then remove “~” from file name). Telephones  
without a local directory, such as new units from the fac-  
tory, 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.  
Central  
(boot  
server)  
•
For information on file format, refer to 3.1.17.1  
Local Contact Directory File Format on page 39.  
XML file: <Ethernet  
This file can be created manually using an XML editor.  
address>-directory.xml  
•
For information on file format, refer to 3.1.17.1  
Local Contact Directory File Format on page 39.  
Web Server  
(if enabled)  
None.  
Local Phone User  
Interface  
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.  
Local  
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3.1.17.1 Local Contact Directory File Format  
An example local contact directory is shown. Look to the table for an explanation of  
each element.  
Local Contact Directory File example:  
<?xml version="1.0" encoding="UTF-8" standalone="yes" ?>  
<directory>  
<item_list>  
<item>  
<ln>Doe</ln>  
<fn>John</fn>  
<ct>1001</ct>  
<sd>1</sd>  
<rt>1</rt>  
<dc />  
<ad>0</ad>  
<ar>0</ar>  
<bw>0</bw>  
<bb>0</bb>  
</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  
first name  
a
40 bytes  
ln  
UTF-8 encoded string of up to  
40 bytes  
last name  
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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  
contact  
Cannot be Null or duplicated; is used by the phone  
to address a remote party in the same way that a  
string of digits or a SIP URL are dialed manually  
by the user. This element is also used to associate  
incoming callers with a particular directory entry.  
sd  
rt  
Null, 1 to 9999  
Null, 1 to 21  
speed-dial index  
Associates a particular entry with a speed dial bin  
for one-touch dialing or dialing from the speed dial  
menu.  
ring type  
When incoming calls can be associated with a  
directory entry by matching the address fields, this  
field is used to specify ring type to be used.  
dc  
ad  
UTF-8 encoded string contain-  
ing digits (the user part of a SIP  
URL) or a string that constitutes  
a valid SIP URL  
divert contact  
The forward-to address for the autodivert feature.  
0,1  
auto divert  
If 1, automatically diverts callers that match the  
directory entry to the address specified in divert-  
contact.  
b
ar  
0,1  
auto-reject  
If 1, automatically rejects callers that match the  
directory entry.  
bw  
bb  
0,1  
0,1  
buddywatching  
If 1, add this contact to the list of watched phones.  
buddyblock  
If 1, block this contact from watching this phone.  
a. In some cases, this will be less than 40 characters due to UTF-8’s variable length encoding.  
b. If auto-divert is also enabled, it has precedence over auto-reject.  
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 simi-  
lar to the digit map feature of the Media Gateway Control Protocol (MGCP) and the  
40  
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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.  
Configuration file:  
sip.cfg  
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.  
Central  
(boot  
server)  
Configuration file:  
phone1.cfg  
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.  
Web Server  
(if enabled)  
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.  
Local  
Local Phone User  
Interface  
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.  
Configuration file:  
phone1.cfg  
Specify the number of line keys to assign per registration.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.2.1 Registration  
<reg/> on page 149.  
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Features  
Web Server  
(if enabled)  
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 per-  
manently override global settings unless deleted through  
the Reset Local Config menu selection.  
Local  
Local Phone User  
Interface  
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 Configura-  
tion 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).  
Configuration file:  
sip.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.  
Central  
(boot  
server)  
Configuration file:  
phone1.cfg  
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.  
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Features  
Web Server  
(if enabled)  
Specify the default number of calls that can be active or  
on hold per line key and the number of calls per registra-  
tion 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 per-  
manently override global settings unless deleted through  
the Reset Local Config menu selection.  
Local  
Local Phone User  
Interface  
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 Configura-  
tion 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 appear-  
ance. 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 con-  
trol 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).  
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Configuration file:  
sip.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.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.1.4.4 Special  
Events <specialEvent/> on page 93.  
Configuration file:  
phone1.cfg  
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 dis-  
abled on shared lines.  
•
For more information, refer to 4.6.2.3 Diversion  
<divert/> on page 154.  
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Features  
Web Server  
(if enabled)  
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. Nav-  
igate 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:  
Local  
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 per-  
manently override global settings unless deleted through  
the Reset Local Config menu selection.  
Local Phone User  
Interface  
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-  
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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.  
Configuration file:  
sip.cfg  
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.  
Configuration file:  
phone1.cfg  
Specify per-registration line type (private or shared) and  
the shared line third party name. A shared line will sub-  
scribe to a server providing call state information.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.2.1 Registration  
<reg/> on page 149.  
Specify per-registration whether diversion should be dis-  
abled on shared lines.  
•
For more information, refer to 4.6.2.3 Diversion  
<divert/> on page 154.  
Web Server  
(if enabled)  
Specify per-registration line type (private or shared) and  
third party name, and whether diversion should be dis-  
abled 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 per-  
manently override global settings unless deleted through  
the Reset Local Confide menu selection.  
Local  
Local Phone User  
Interface  
Specify per-registration line type (private or shared) and  
the shared line third party name using the SIP Configura-  
tion menu. Either the Web Server or the boot server con-  
figuration files or the local phone user interface should be  
used to configure registrations, not a mixture of these  
options. When the SIP Configuration menu is used, it is  
assumed that all registrations use the same server.  
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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).  
Configuration file:  
sip.cfg  
None.  
Central  
(boot  
server)  
Configuration file:  
phone1.cfg  
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.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
3.1.25 Customizable Fonts and Indicators  
The phone’s user interface can be customized by changing the fonts and graphic icons  
used on the display and the LED indicator patterns. Pre-existing fonts embedded in the  
software can be overwritten or new fonts can be downloaded. The bitmaps and bitmap  
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animations used for graphic icons on the display can be changed and repositioned.  
LED flashing sequences and colors can be changed.  
Configuration File: Specify fonts to overwrite existing ones or specify new fonts.  
sip.cfg  
•
For more information, refer to 4.6.1.15 Fonts <font/> on  
page 129.  
Specify which bitmaps to use.  
Central  
(boot  
server)  
•
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 <indica-  
tors/> on page 134.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
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  
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shown on the idle display if the SIP server supports this feature. Refer to 3.4.1 Pres-  
ence on page 60.  
XML file:  
The <sd>x</sd> element in the <Ethernet  
<Ethernet address>-directory.xml address>-directory.xml file links a directory  
entry to a speed dial resource within the  
phone. Speed dial entries are mapped auto-  
matically to unused line keys (line keys are  
not available on the IP 4000) and are avail-  
able for selection within the speed dial menu.  
(Press the up-arrow key from the idle display  
to jump to SpeedDial).  
Central  
(boot  
server)  
•
For more information, refer to 3.1.17.1  
Local Contact Directory File Format on  
page 39.  
Web Server (if enabled)  
None.  
Local Phone User Interface  
The next available Speed Dial Index is  
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.  
Local  
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  
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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.  
Configuration file:  
sip.cfg  
Turn time and date display on or off.  
•
For more information, refer to 4.6.1.4 User Prefer-  
ences <user_preferences/> on page 98.  
Set the time and date display formats.  
Central  
(boot  
server)  
•
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 param-  
eters.  
•
For more information, refer to 4.6.1.10.2 Time Syn-  
chronization <SNTP/> on page 121.  
Web Server  
(if enabled)  
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.  
Local Phone User  
Interface  
The basic SNTP settings can be made in the Network  
Configuration menu.  
Local  
•
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.  
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3.1.29 Idle Display Animation  
All phones except the SoundPoint® IP 300 and SoundPoint® IP 301 can display a cus-  
tomized animation on the idle display in addition to the time and date. For example, a  
company logo could be displayed.  
Configuration file:  
sip.cfg  
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 Anima-  
tions <Animations/> <IP_300/>, <IP_400/>,  
<IP_500/>, <IP_600/> and <IP_4000/> on  
page 134.  
Central  
(boot  
server)  
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.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
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.  
Configuration file:  
phone1.cfg  
Specify which registrations have the feature and what  
contact to call when going off hook.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.2.2.2 Automatic  
Off-hook Call Placement <autoOffHook/> on  
page 153.  
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Web Server  
None.  
None.  
(if enabled)  
Local  
Local Phone User  
Interface  
3.2.2 Call Hold  
Call hold is a fundamental feature of the phone. The purpose of hold is to pause activ-  
ity 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.  
Configuration file: Specify whether RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sen-  
sip.cfg  
donly or a=inactive) outgoing hold signaling is used.  
•
For more information, refer to 4.6.1.1.4 SIP <SIP/> on  
page 88.  
Central  
(boot  
server)  
Specify local hold reminder options.  
•
For more information, refer to 4.6.1.12.2 Hold, Local  
Reminder <hold/><localReminder/> on page 127.  
Web Server  
(if enabled)  
Specify whether or not to use RFC 2543 (c=0.0.0.0) outgo-  
ing 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 perma-  
nently override global settings unless deleted through the  
Reset Local Config menu selection.  
Local  
Local Phone User  
Interface  
Use the SIP Configuration menu to specify whether or not to  
use RFC 2543 (c=0.0.0.0) outgoing hold signaling. The  
alternative is RFC 3264 (a=sendonly or a=inactive).  
3.2.3 Call Transfer  
Call transfer enables the user (User A or transferring user) to transform an existing call  
with User B (primary call) into a new call between User B and a third user C (trans-  
ferred-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.  
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• 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.  
Configuration file: Specify whether to allow a transfer during the proceeding  
Central  
(boot  
server)  
sip.cfg  
state of a consultation call.  
•
For more information, refer to 4.6.1.1.4 SIP <SIP/> on  
page 88.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
3.2.4 Three-Way Conference, Local or Centralized  
2
Local or centralized conferences 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.  
Configuration file: Specify which type of conference to establish and the  
Central  
(boot  
server)  
sip.cfg  
address of the centralized conference resource.  
•
For more information, refer to 4.6.1.1.4.5 Conference  
Setup <conference/> on page 93.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
®
2. On SoundStation IP 4000, conferences are not available if the G.729 codec is enabled on the phone.  
This restriction will be removed in future releases.  
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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 spe-  
cific 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.  
Configuration file: Set all call diversion settings including a global forward-to  
phone1.cfg  
contact and individual settings for call forward all, call for-  
ward busy, call forward no-answer, and call forward do-not-  
disturb.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.2.3 Diversion  
<divert/> on page 154.  
Web Server  
(if enabled)  
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 perma-  
nently override global settings unless deleted through the  
Reset Local Config menu selection.  
Local  
Local Phone User  
Interface  
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.  
Changes are saved to local flash and backed up to <Ethernet  
address>-phone.cfg on the boot server. They will perma-  
nently override global settings unless deleted through the  
Reset Local Config menu selection.  
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.  
Configuration file:  
sip.cfg  
Turn this feature on or off.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.24 Feature <fea-  
ture/> on page 144.  
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Web Server  
None.  
None.  
(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.  
Configuration file:  
sip.cfg  
Turn this feature on or off.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.24 Feature <fea-  
ture/> on page 144.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
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.  
Configuration file:  
sip.cfg  
Turn this feature on or off.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.24 Feature <fea-  
ture/> on page 144.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
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3.2.9 Last Call Return  
The phone allows server-based last call return. This feature depends on support from a  
SIP server.  
Configuration file:  
sip.cfg  
Turn this feature on or off.  
•
For more information, refer to 4.6.1.24 Feature <fea-  
ture/> on page 144.  
Central  
(boot  
server)  
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.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
3.3 Audio Processing Features  
Proprietary state-of-the-art digital signal processing (DSP) technology is used to pro-  
vide 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 sys-  
tem 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-  
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urable for different network environments. When packets are lost, a concealment algo-  
rithm minimizes the resulting negative audio consequences.  
Configuration file: Set the jitter buffer tuning parameters including minimum  
Central  
(boot  
server)  
sip.cfg  
and maximum size and shrink aggression.  
•
For more information, refer to 4.6.1.8.1.2 Codec Pro-  
files <profiles/> on page 109.  
Web Server  
(if enabled)  
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 perma-  
nently override global settings unless deleted through the  
Reset Local Config menu selection.  
Local  
Local Phone User  
Interface  
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 pack-  
ets, efficiently regenerating a facsimile of the background noise at the remote end.  
Configurationfile: Enable or disable VAD and set the detection threshold.  
Central  
sip.cfg  
(boot  
•
For more information, refer to 4.6.1.8.10 Voice Activity  
Detection <VAD/> on page 118.  
server)  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
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3.3.4 DTMF Tone Generation  
The phone generates dual tone multi-frequency (DTMF) tones in response to user dial-  
ing 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 deci-  
sion is based on the capabilities of the remote end point.  
Configuration file: Set the DTMF tone levels, autodialing on and off times, and  
Central  
(boot  
server)  
sip.cfg  
other parameters.  
For more information, refer to 4.6.1.5.1 Dual Tone  
Multi-Frequency <DTMF/> on page 99.  
•
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
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.  
Configuration file: Enable or disable RFC 2833 support in SDP offers and spec-  
Central  
(boot  
server)  
sip.cfg  
ify 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)  
None.  
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.  
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3.3.7 Audio Codecs  
The following table summarizes the phone’s audio codec support:  
Effective  
audio band-  
width  
Sample  
Rate  
Algorithm  
MIME Type  
Ref.  
Bit Rate  
Frame Size  
G.711μ-law  
PMCU  
RFC  
1890  
64 Kbps  
8 Ksps  
10ms - 80ms  
3.5KHz  
3.5KHz  
3.5KHz  
N/A  
G.711a-law  
G.729AB  
SID  
PCMA  
G729  
RFC  
1890  
64 Kbps  
8 Kbps  
N/A  
8 Ksps  
8 Ksps  
N/A  
10ms - 80ms  
10ms - 80ms  
N/A  
RFC  
1890  
CN  
RFC  
3389  
RFC 2833  
phone-event  
RFC  
2833  
N/A  
N/A  
N/A  
N/A  
Configuration file:  
sip.cfg  
Specify codec priority, preferred payload sizes, and jitter  
buffer tuning parameters.  
Central  
(boot  
server)  
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.  
Web Server  
(if enabled)  
Specify codec priority, preferred payload sizes, and jitter  
buffer tuning parameters.  
Navigate to: http://<phoneIPAddress>/coreConf.htm#au  
Changes are saved to local flash and backed up to <Ethernet  
address>-phone.cfg on the boot server. Changes will per-  
manently override global settings unless deleted through the  
Reset Local Config menu selection.  
Local  
Local Phone User  
Interface  
None.  
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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 fea-  
ture 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  
3
the local talker in certain circumstances. 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  
4
notified when a change in monitored status occurs . Phone status changes are broad-  
cast automatically to monitoring phones when the user engages in calls or invokes do-  
3. AGC support will be available in a subsequent release.  
4. Notification when a change in monitored status occurs will be available in a subsequent release.  
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not-disturb. The user can also manually specify a state to convey, overriding, and per-  
haps masking, the automatic behavior.  
XML file: <Ethernet  
address>-direc-  
tory.xml  
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.  
Central  
(boot  
server)  
•
For more information, refer to 3.1.17.1 Local Contact  
Directory File Format on page 39.  
Web Server  
(if enabled)  
None.  
Local Phone User  
Interface  
The user can edit the directory contents. The Watch  
Buddy and Block Buddy fields control the buddy behavior  
of contacts.  
Local  
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.  
3.4.2 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 con-  
tact directory, the preferred method.  
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  
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only on the SoundPoint® IP 600’s and 601’s and SoundStation® IP 4000’s higher reso-  
lution displays.  
Basic character support includes the following Unicode character ranges:  
Name  
Range  
C0 Controls and Basic Latin  
C1 Controls and Latin-1 Supplement  
Cyrillic (partial)  
U+0000 - U+007F  
U+0080 - U+00FF  
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  
Hiragana  
U+3000 - U+303F  
U+3040 - U+309F  
U+30A0 - U+30FF  
U+3100 - U+312F  
U+3130 - U+318F  
U+31A0 - U+31BF  
U+3200 - U+327F  
U+3300 - U+33FF  
U+4E00 - U+9FFF  
U+AC00 - U+D7A3  
U+F900 - U+FAFF  
U+FF00 - U+FFFF  
Katakana  
Bopomofo  
Hangul Compatibility Jamo  
Bopomofo Extended  
Enclosed CJK Letters and Months  
CJK Compatibility  
CJK Unified Ideographs  
Hangul Syllables  
CJK Compatibility Ideographs  
CJK Half-width forms  
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 inaccessi-  
ble, the language will revert to the internal language.  
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Note  
Currently, the multilingual feature is only available in the application. At this time, the bootROM  
application is English only.  
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.  
Web Server  
(if enabled)  
None.  
Local Phone User  
Interface  
The user can select the preferred language under the Set-  
tings 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.  
Local  
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-  
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ing the life cycle of a call. These call progress tones are easily configurable for com-  
patibility with worldwide telephony standards or local preferences.  
Configuration file:  
sip.cfg  
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.  
Central  
(boot  
server)  
•
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.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
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 config-  
ured 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,  
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600 and 601). A message-waiting signal from a voice mail server will trigger the mes-  
sage-waiting indicator to flash.  
Configuration file:  
sip.cfg  
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.  
Configuration file:  
phone1.cfg  
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  
Central  
(boot  
server)  
®
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 call-  
back or another contact), and the contact to call when the  
user accesses voice mail.  
•
For more information, refer to 4.6.2.5 Messaging <msg/  
> on page 159.  
Web Server  
(if enabled)  
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).  
Navigate to: http://<phoneIPAddress>/coreConf.htm#us  
On a per-registration basis, specify a subscribe contact for  
solicited NOTIFY applications, a callback mode (self call-  
back or another contact) to call when the user accesses voice  
mail.  
Local  
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.  
Local Phone User  
Interface  
None.  
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3.6.2 Multiple Registrations  
SoundPoint® IP phones support multiple registrations per phone and the SoundSta-  
tion® IP 4000 supports a single registration. The SoundPoint® IP 300 and 301 support  
a maximum of two registrations, the SoundPoint® IP 430 supports two, the Sound-  
Point® 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 registra-  
tions.  
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.  
Configuration file: Specify the local SIP signaling port and an array of SIP serv-  
sip.cfg  
ers 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.  
Configuration file: For up to twelve registrations, specify a display name, a SIP  
phone1.cfg  
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 serv-  
ers and their associated parameters will override the servers  
specified in sip.cfg if non-Null.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.2.1 Registration <reg/  
> on page 149.  
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Web Server  
(if enabled)  
Specify the local SIP signaling port and an array of SIP serv-  
ers 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 app-  
Conf.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 perma-  
nently override global settings unless deleted through the  
Reset Local Config menu selection.  
Local  
Local Phone User  
Interface  
Use the SIP Configuration menu to specify the local SIP sig-  
naling port, a default SIP server to register to and registra-  
tion 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 registra-  
tions use the same server.  
Changes are saved to local flash and backed up to <Ethernet  
address>-phone.cfg on the boot server. Changes will perma-  
nently 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.  
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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.  
Configuration file:  
sip.cfg  
Turn this feature on or off.  
•
For more information, refer to 4.6.1.24 Feature <fea-  
ture/> on page 144.  
Central  
(boot  
server)  
Configuration file:  
phone1.cfg  
Enable this feature per registration.  
•
For more information, refer to 4.6.2.1 Registration  
<reg/> on page 149.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
3.6.4 ACD agent available / unavailable  
The phone supports ACD (Automatic Call Distribution) agent available and unavail-  
able. This feature depends on support from a SIP server.  
Configuration file:  
sip.cfg  
Turn this feature on or off.  
•
For more information, refer to 4.6.1.24 Feature <fea-  
ture/> on page 144.  
Central  
(boot  
server)  
Configuration file:  
phone1.cfg  
Enable this feature per registration.  
•
For more information, refer to 4.6.2.1 Registration  
<reg/> on page 149.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
3.6.5 Server Redundancy  
The phone can be configured with multiple SIP servers, one primary and one or more  
backup. The phone will switch to a backup server when the current primary server  
fails. Backup server configuration can be static or can use advanced DNS methods. In  
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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 mak-  
ing 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  
to 4.6.1.1.2 Server <server/> on page 85 and 4.6.2.1 Registration <reg/> on  
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 2005 Integration  
®
®
SoundPoint IP phones can used with Microsoft Office Live Communications  
®
Server 2005 and Microsoft Office Communicator to help improve business efficien-  
cies and increase productivity and to share ideas and information immediately with  
business contacts.  
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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).  
®
Configuration file:  
sip.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 SUB-  
SCRIBE for presence.  
•
For more information, refer to 4.6.1.14 Presence <pres-  
ence/> on page 129.  
Turn the presence and messaging features on or off.  
•
For more information, refer to 4.6.1.24 Feature <fea-  
ture/> on page 144.  
Central  
(boot  
server)  
Configuration file: Specify the number of line keys to assign per registration.  
phone1.cfg  
•
For more information, refer to 4.6.2.1 Registration <reg/  
> on page 149  
Specify the line/registration number which has roaming bud-  
dies 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 pri-  
vacy support enabled.  
•
For more information, refer to 4.6.2.9 Roaming Privacy  
<roaming_privacy/> on page 162.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
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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 regis-  
tration to call server—that is not Live Communications Server (LCS)—and a second-  
ary 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/> on page 161.  
10. Locate the roaming_privacy attribute.  
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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 Communica-  
tions 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/> on 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.  
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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.  
Configuration file: Specify the Services browser home page, a proxy to use, and  
Central  
(boot  
server)  
sip.cfg  
size limits.  
For more information, refer to 4.6.1.26 MicroBrowser  
<microbrowser/> on page 146.  
•
Web Server  
(if enabled)  
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 perma-  
nently override global settings unless deleted through the  
Reset Local Config menu selection.  
Local  
Local Phone User  
Interface  
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 adminis-  
trator, 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.  
Configuration file:  
sip.cfg  
Specify the minimum lengths for the user and administrator  
passwords.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.20.2 Password  
Lengths <pwd/><length/> on page 141.  
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Web Server  
(if enabled)  
None.  
Local Phone User  
Interface  
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 charac-  
ters 32-127 (0x20-0x7F) only.  
Local  
Changes are saved to local flash but are not backed up to  
<Ethernet address>-phone.cfg on the boot server for secu-  
rity reasons.  
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 cer-  
tificate.  
Central  
(boot  
Configuration file:  
None.  
None.  
server)  
Web Server  
(if enabled)  
Local  
Local Phone User  
Interface  
The custom certificate can be specified and the type of cer-  
tificate 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 signal-  
ing:  
• source IP address validation  
• digest authentication  
• both  
Configuration File: Specify the type of validation to perform on a request-by-  
sip.cfg  
request basis, appropriate to specific event types in some  
cases.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.1.4.3 Request Vali-  
dation <requestValidation/> on page 92.  
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Features  
Web Server (if  
enabled)  
None.  
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 con-  
figuration 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), con-  
tact 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.  
Configuration File: Specify the phone-specific contact directory and the phone-  
sip.cfg  
specific configuration override file.  
•
For more information, refer to section 4.6.1.20.1  
Encryption <encryption/> on page 141.  
Central  
(boot  
server)  
Configuration file:  
<device>.cfg  
Change the encryption key.  
•
For more information, refer to section 2.2.2.1.1.3 Set-  
ting Flash Parameters from Configuration Files on  
page 16.  
Web Server (if  
enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
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Note  
®
The SoundPoint IP 300 and 500 phones will always fail at decrypting files. These phones will recog-  
nize 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 301, 430, 501, 600, and 601 and the SoundStation  
®
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.  
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4 Optimization  
4.1 Ethernet Switch  
The SoundPoint® IP phones contain two Ethernet ports, labeled LAN and PC, and an  
embedded Ethernet switch that runs at full line-rate. The Ethernet switch allows a per-  
sonal 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® IP switch gives higher transmit priority to packets originating in the  
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.  
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.  
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The phone can filter incoming RTP packets arriving on a particular port by IP address.  
Packets arriving from a non-negotiated IP address can be discarded.  
The phone can also enforce symmetric port operation for RTP packets: packets arriv-  
ing with the source port set to other than the negotiated remote sink port can be  
rejected.  
The phone can also jam the destination transport port to a specified value regardless of  
the negotiated port. This can be useful for punching through firewalls. When this is  
enabled, all RTP traffic will be sent to the specified port and will be expected to arrive  
on that port as well. Incoming packets are sorted by the source IP address and port,  
allowing multiple RTP streams to be multiplexed.  
The RTP port range used by the phone can be specified. Since conferencing and multi-  
ple 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  
Specify whether to filter incoming RTP packets by IP  
address, whether to require symmetric port usage, whether  
to jam the destination port and specify the local RTP port  
range start.  
Central  
(boot  
server)  
•
For more information, refer to 4.6.1.10.3.1 RTP <RTP/  
> on page 124.  
Web Server  
(if enabled)  
Specify whether to filter incoming RTP packets by IP  
address, whether to require symmetric port usage, whether  
to jam the destination port and specify the local RTP port  
range start.  
Navigate to: http://<phoneIPAddress>/netConf.htm#rt  
Local  
Changes are saved to local flash and backed up to <Ethernet  
address>-phone.cfg on the boot server. They will perma-  
nently override global settings unless deleted through the  
Reset Local Config menu selection.  
Local Phone User  
Interface  
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 config-  
ured on a per-phone basis.  
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Configuration file: Specify the external NAT IP address and the ports to be used  
Central  
(boot  
server)  
phone1.cfg  
for signaling and RTP traffic.  
•
For more information, refer to 4.6.2.6 Network Address  
Translation <nat/> on page 160.  
Web Server  
(if enabled)  
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 perma-  
nently override global settings unless deleted through the  
Reset Local Config menu selection.  
Local  
Local Phone User  
Interface  
None.  
4.3 Updating and Rebooting  
The bootROM, application executable, and configuration files can be updated auto-  
matically 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.  
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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:  
®
Volume-, Volume+, Hold, Do Not Disturb  
Volume-, Volume+, Hold, Messages  
Volume-, Volume+, Mute, Messages  
*, #, Volume+, Select  
SoundPoint IP 300 and 301:  
®
SoundPoint IP 430, 500, and 501:  
®
SoundPoint IP 600 and 601:  
®
SoundStation IP 4000:  
Centralized Reboot  
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  
The phones maintain both boot and application event log files. These files can be help-  
ful 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-  
5
urable directory on the boot server. Both overwrite and append modes are supported  
for the application event log.  
The event log files are:  
• <Ethernet address>-boot.log  
• <Ethernet address>-app.log  
The boot log file is uploaded to the boot server after every reboot.  
5. HTTP and TFTP don’t support append mode unless server settings are changed for this.  
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The application log file is uploaded periodically or when the local copy reaches a pre-  
determined 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:  
®
Line1, Line2, Arrow Up, Arrow Down  
The four arrow keys  
SoundPoint IP 300 and 301:  
®
SoundPoint IP 430, 500,  
502, 600, and 601:  
®
Menu, Exit, Off-hook/Hands-free, Redial  
SoundStation IP 4000:  
Log files uploaded in this manner are named:  
• <Ethernet address>-now-boot.log  
• <Ethernet address>-now-app.log  
Configuration file:  
sip.cfg  
Specify a multitude of event logging settings.  
•
For more information, refer to 4.6.1.19 Event Logging  
<logging/> on page 137.  
Central  
(boot  
server)  
Configuration file:  
<Ethernet  
address>.cfg  
Specify different directory to use for log files if desired.  
•
For more information, refer to 2.2.2.1.1.1 Master Con-  
figuration Files on page 13.  
Web Server  
(if enabled)  
Specify a multitude of event logging settings.  
Navigate to: http://<phoneIPAddress>/coreConf.htm#lo  
Local  
Local Phone User  
Interface  
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  
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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.  
Configuration file: Specify protocol-specific IP TOS settings.  
Central  
sip.cfg  
(boot  
•
For more information, refer to 4.6.1.9.2 IP TOS <IP/> on  
page 119.  
server)  
Web Server  
(if enabled)  
Specify IP TOS settings.  
Navigate to: http://<phoneIPAddress>/netConf.htm#qo  
Local  
Local Phone User  
Interface  
None.  
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.  
Configuration file: Specify default and protocol-specific 802.1p/Q settings.  
Central  
sip.cfg  
(boot  
•
For more information, refer to 4.6.1.9.1 Ethernet IEEE  
802.1p/Q <Ethernet/> on page 118.  
server)  
Web Server  
(if enabled)  
Specify 802.1p/Q settings.  
Navigate to http://<phoneIPAddress>/netConf.htm#qo  
Local Phone User  
Interface  
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 auto-  
boot countdown.  
Local  
Phase 2: Application - Navigate to: Menu>Set-  
tings>Advanced>Admin Settings>Network Configuration  
•
For more information, refer to 2.2.1 Basic Network Setup  
on page 5.  
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4.5.3 RTCP Support  
The phone supports RTCP per RFC 1889. For each RTP stream, which, by conven-  
tion, uses even ports only, the next higher odd port is used to send and receive RTCP  
reports.  
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4.6 Configuration Files  
This section is a reference for all parameters that are configurable when using the cen-  
tralized 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  
&
”
&amp;  
&quot;  
&apos;  
&lt;  
’
<
>
&gt;  
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 sec-  
tions 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.  
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4.6.1.1 Protocol <volpProt/>  
4.6.1.1.1 Local <local/>  
Permitted  
Values  
Attribute  
Default Interpretation  
volpProt.local.port  
0 to 65535  
5060  
Local port for sending and receiving SIP signaling  
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.  
4.6.1.1.2 Server <server/>  
Permitted  
Values  
Attribute  
Default Interpretation  
voIpProt.server.dhcp.available  
0, 1  
0
If set to 1, check with the  
DHCP server for SIP server IP  
address. If set to 0, do not  
check with DHCP server.  
voIpProt.server.dhcp.option  
voIpProt.server.dhcp.type  
128 to 255  
Option to request from the  
DHCP server if voIp-  
Prot.server.dhcp.available = 1.  
There is no default value for  
this parameter, it must be filled  
in with a valid value.  
0, 1  
If set to 0, IP request address.  
If set to 1, request string.  
Type to request from the  
DHCP server if voIp-  
Prot.server.dhcp.available = 1.  
There is no default value for  
this parameter, it must be filled  
in with a valid value.  
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Permitted  
Values  
Attribute  
Default Interpretation  
voIpProt.server.x.address  
dotted-decimal  
IP address or  
host name  
Null  
IP address or host name and  
port of a SIP server that accepts  
registrations. Multiple servers  
can be listed starting with x=1,  
2, ... for fault tolerance.  
voIpProt.server.x.port  
0, Null, 1 to  
65535  
Null  
If port is 0 or Null:  
If voIpProt.server.x.address is  
a hostname and voIp-  
Prot.server.x.transport is set to  
DNSnaptr, do NAPTR then  
SRV lookups.  
If voIpProt.server.x.transport is  
set to TCPpreferred or  
UDPonly then use 5060 and  
don’t advertise the port number  
in signalling.  
If voIpProt.server.x.address is  
an IP address, there is no DNS  
lookup and 5060 is used for the  
port but it is not advertised in  
signaling.  
If port is 1 to 65535:  
This value is used and it is  
advertised in signaling.  
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Permitted  
Values  
Attribute  
Default Interpretation  
voIpProt.server.x.transport  
DNSnaptr or  
DNSna If set to Null or DNSnaptr:  
TCPpreferred or ptr  
UDPonly or  
TLS  
If voIpProt.server.x.address is  
a hostname and voIp-  
Prot.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 voIp-  
Prot.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 IP 300 and 500  
phones.  
voIpProt.server.x.expires  
positive integer,  
minimum 300  
3600  
Requested registration period  
a
in seconds .  
voIpProt.server.x.expires.overlap  
positive integer,  
minimum 5,  
maximum  
60  
The number 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  
65535  
returned by the server if that  
value is less than the config-  
ured overlap value.  
voIpProt.server.x.register  
0, 1  
1
0
If set to 0, calls can be routed to  
an outbound proxy without reg-  
istration.  
voIpProt.server.x.retryTimeOut  
Null or  
non-negative  
integer  
If set to 0 or Null, use standard  
RFC 3261 signaling retry  
behavior. Otherwise retryTim-  
eOut determines how often  
retries will be sent.  
Units = milliSeconds. (Finest  
resolution = 100ms).  
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Permitted  
Values  
Attribute  
Default Interpretation  
voIpProt.server.x.retryMaxCount  
Null or  
3
If set to 0 or Null, 3 is used.  
non-negative  
integer  
retryMaxCount retries will be  
attempted before moving on to  
the next available server.  
voIpProt.server.x.expires.lineSeize  
positive integer,  
minimum 10  
30  
Requested line-seize subscrip-  
tion period.  
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/>  
Permitted  
Values  
Attribute  
Default Interpretation  
volpProt.SDP.answer.userLo-  
calPreferences  
0 or 1  
0
If set to 1, the phones uses its own pref-  
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/>  
Permitted  
Values  
Attribute  
Default Interpretation  
voIpProt.SIP.useRFC2543hold  
0, 1  
0
If set to 1, use the obsolete c=0.0.0.0  
RFC2543 technique, otherwise, use  
SDP media direction attributes (such as  
a=sendonly) per RFC 3264 when initi-  
ating hold. In either case, the phone  
processes incoming hold signaling in  
either format.  
voIpProt.SIP.lcs  
0, 1  
0
If set to 1, the proprietary “epid”  
parameter is added to the From field of  
®
all requests to support Microsoft  
Office Live Communications  
Server 2005.  
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Permitted  
Values  
Attribute  
Default Interpretation  
voIpProt.SIP.ms-forking  
0, 1  
0
If set to 0, support for MS-forking is  
disabled. If set to 1, support for MS-  
forking is enabled and the phone will  
reject all Instant Message INVITEs.  
This parameter is relevant for  
®
Microsoft Office Live Communica-  
tions Server 2005 server installations.  
Note that if any end point registered to  
the same account has MS-forking dis-  
abled, all other end points default back  
®
to non-forking mode. Windows Mes-  
senger does not use MS-forking so be  
aware of this behavior if one of the end  
®
points is Windows Messenger.  
voIpProt.SIP.dialog.usePvalue  
0, 1  
0
If set to 0, phone uses "pval" field  
name in Dialog. This obeys the draft-  
ietf-sipping-dialog-package-06.txt  
draft. If set to 1, phone uses a field  
name of "pvalue".  
voIpProt.SIP.connection-  
Reuse.useAlias  
0, 1  
0, 1  
0
0
If set to 0, shows old behavior.  
If set to 1, phone uses the connection  
reuse draft which introduces "alias".  
voIpProt.SIP.sendCompactHdrs  
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.session-  
Timers  
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.request-  
URI.E164.addGlobalPrefix  
0, 1  
0
If set to 1, ‘+’ global prefix is added to  
E.164 user parts in sip: URIs:.  
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Permitted  
Values  
Attribute  
Default Interpretation  
voIpProt.SIP.allowTransferOn-  
Proceeding  
0, 1  
1
If set to 1, a transfer can be completed  
during the proceeding state of a consul-  
tation call. This is the default.  
If set to 0, a transfer is not allowed dur-  
ing the proceeding state of a consulta-  
tion call.  
voIpProt.SIP.dialog.useSDP  
voIpProt.SIP.pingInterval  
0, 1  
0
0
If set to 0, new dialog event package  
draft is used (no SDP in dialog body).  
If set to 1, for backwards compatibility,  
use this setting to send SDP in dialog  
body.  
0 to 3600  
The number in seconds to send "PING"  
message. This feature is disabled by  
default.  
4.6.1.1.4.1 Outbound Proxy <outboundProxy/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
voIpProt.SIP.outboundProxy.address  
dotted-deci-  
mal IP address  
or host name  
Null  
IP address or host name and  
port of a SIP server to which  
the phone shall send all  
requests.  
voIpProt.SIP.outboundProxy.port  
1 to 65535  
5060  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
voIpProt.SIP.outboundProxy.transport DNSnaptr or  
DNSnap If set to Null or DNSnaptr:  
TCPpreferred  
or  
UDPonly or  
TLS  
tr  
If voIpProt.SIP.outbound-  
Proxy.address is a hostname  
and voIpProt.SIP.outbound-  
Proxy.port is 0 or Null, do  
NAPTR then SRV look-ups  
to try to discover the trans-  
port, ports and servers, as  
per RFC 3263. If voIp-  
Prot.SIP.outbound-  
Proxy.address is an IP  
address, or a port is given,  
then UDP is used.  
If set to TCPpreferred:  
TCP is the preferred trans-  
port, 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 IP 300 and  
500 phones.  
4.6.1.1.4.2 Alert Information <alertInfo/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
volpProt.SIP.alertInfo.x.value  
string to com-  
pare against  
the value of  
Alert-Info  
headers in  
INVITE  
Null  
Alert-Info fields from  
INVITE requests will be  
compared against as many of  
these parameters as are spec-  
ified (x=1, 2, ..., N) and if a  
match is found, the behavior  
described in the correspond-  
ing ring class (refer to  
4.6.1.7.2 Ring type <ring-  
Type/> on page 107) will be  
applied.  
requests  
voIpProt.SIP.alertInfo.x.class  
positive  
integer  
Null  
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4.6.1.1.4.3 Request Validation <requestValidation/>  
Attribute  
Permitted Values Default  
Interpretation  
voIpProt.SIP.requestValida-  
tion.x.request  
One of:  
Null  
Sets the name of the method  
for which validation will be  
applied.  
“INVITE”,  
“ACK”, “BYE”,  
“REGISTER”,  
“CANCEL”,  
“OPTIONS”,  
“INFO”,  
“MESSAGE”,  
“SUB-  
SCRIBE”,  
WARNING: Intensive  
request validation may have a  
negative performance impact  
due to the additional signal-  
ing required in some cases,  
therefore, use it judiciously.  
“NOTIFY”,  
“REFER”,  
“PRACK”, or  
“UPDATE”  
voIpProt.SIP.requestValida-  
tion.x.method  
Null or  
Null  
If Null, no validation is done.  
Otherwise this sets the type of  
validation performed for the  
request:  
one of: “source”,  
“digest” or  
“both”/”all”  
source: ensure request is  
received from an IP address  
of a server belonging to the  
set of target registration serv-  
ers;  
digest: challenge requests  
with digest authentication  
using the local credentials for  
the associated registration  
(line);  
both or all: apply both of the  
above methods  
voIpProt.SIP.requestValida-  
tion.x.request.y.event  
A valid string  
Null  
Determines which events  
specified with the Event  
header should be validated;  
only applicable when voIp-  
Prot.SIP.requestValida-  
tion.x.request is set to  
“SUBSCRIBE” or  
“NOTIFY”.  
If set to Null, all events will  
be validated.  
voIpProt.SIP.requestValida-  
tion.digest.realm  
A valid string  
PolycomSPIP Determines string used for  
Realm.  
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4.6.1.1.4.4 Special Events <specialEvent/>  
Permitted  
Optimization  
Attribute  
Values  
Default  
Interpretation  
voIpProt.SIP.specialEv-  
ent.lineSeize.nonStandard  
0, 1  
1
If set to 1, process a 200 OK  
response for a line-seize event  
SUBSCRIBE as though a line-  
seize NOTIFY with Subscription  
State: active header had been  
received, this speeds up process-  
ing.  
voIpProt.SIP.specialEv-  
ent.checkSync.alwaysReboot  
0, 1  
0
If set to 1, always reboot when a  
NOTIFY message is received from  
the server with event equal to  
check-sync.  
If set to 0, only reboot if any of the  
files listed in [mac].cfg have  
changed on the FTP server when a  
NOTIFY message is received from  
the server with event equal to  
check-sync.  
4.6.1.1.4.5 Conference Setup <conference/>  
Permitted Val-  
ues  
Attribute  
Default Interpretation  
voIpProt.SIP.confer-  
ence.address  
ASCII string  
up to 128 char-  
acters long  
Null  
If Null, conferences are set up on the  
phone locally.  
If set to some value, conferences are set  
up by the server using the conferencing  
agent specified by this address. The  
acceptable values depend on the confer-  
encing server implementation policy.  
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4.6.1.2 Dial Plan <dialplan/>  
Permitted  
Optimization  
Attribute  
Values  
Default  
Interpretation  
dialplan.impossibleMatch- 0, 1 or 2  
Handling  
0
If set to 0, the digits entered up to  
and including the point where an  
impossible match occurred are sent  
to the server immediately.  
If set to 1, give reorder tone.  
If set to 2, allow user to accumulate  
digits and dispatch call manually  
with the Send soft key.  
dialplan.removeEndOfDial 0, 1  
1
If set to 1, strip trailing # digit from  
digits sent out.  
4.6.1.2.1 Digit Map <digitmap/>  
Attribute  
Permitted Values  
Default  
Interpretation  
dialplan.digitmap  
string compatible with [2-9]11|0T|  
the digit map feature 011xxx.T|  
of MGCP described in [0-1][2-  
When this attribute is  
present, number-only  
dialing during the setup  
2.1.5 of RFC 3435.  
String is limited to  
512 bytes and 20 seg-  
ments; a comma is  
also allowed; when  
reached in the digit  
map, a comma will  
turn dial tone back on.  
9]xxxxxxxxx|  
[2-9]xxxxxxxxx| be compared against the  
[2-9]xxxT  
phase of new calls will  
patterns therein and if a  
match is found, the call  
will be initiated automat-  
ically eliminating the  
need to press Send.  
dialplan.digitmap.timeOut positive integer  
3
Timeout in seconds for  
‘T’ feature of digitmap.  
4.6.1.2.2 Routing <routing/>  
This configuration section allows the user to create a specific routing path for outgoing  
SIP calls independent of other ‘default’ configuration.  
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4.6.1.2.2.1 Server <server/>  
Attribute  
Permitted Values  
Default  
Interpretation  
dialplan.rout-  
ing.server.x.address  
dotted-decimal IP  
address or host name  
Null  
IP address or host name and  
port of a SIP server that will be  
used for routing calls. Multiple  
servers can be listed starting  
with x=1, 2, ... for fault toler-  
ance.  
dialplan.rout-  
ing.server.x.port  
1 to 65535  
5060  
4.6.1.2.2.2 Emergency <emergency/>  
In the following attributes, x is the index of the emergency entry description and y is  
the index of the server associated with emergency entry x. For each emergency entry  
(index x), one or more server entries (indexes (x,y)) can be configured. x and y must  
both use sequential numbering starting at 1.  
Attribute  
Permitted Values  
Default  
Interpretation  
dialplan.routing.emer-  
gency.x.value  
Comma separated list  
of entries or single  
entry representing a  
SIP URL or a combi-  
nation of SIP URLs.  
Null  
This determines the  
URLs that should be  
watched for.  
Example:  
“15,17,18”,  
“911”, “sos”.  
When one of these  
defined URLs is detected  
as having been dialed by  
the user, the call will  
automatically be directed  
to the defined emergency  
server.  
dialplan.routing.emer-  
gency.x.server.y  
positive integer  
Null  
Index representing the  
server defined in  
4.6.1.2.2.1 Server  
<server/> on page 95  
that will be used for  
emergency routing.  
4.6.1.3 Localization <localization/>  
The phone has a multilingual user interface. It supports both North American and  
international time and date formats. The call progress tones can also be customized.  
For more information, refer to 4.6.1.5.2 Chord Sets <chord_sets/> on page 100, and  
4.6.1.7.1.1 Call Progress Patterns on page 105.  
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4.6.1.3.1 Multilingual <multilingual/>  
The multilingual feature is based on string dictionary files downloaded from the boot  
server. These files are encoded in standalone XML format. Several western European  
and Asian languages are included with the distribution.  
Permitted  
Values  
Attribute  
Interpretation  
lcl.ml.lang  
Null  
OR  
If Null, the default internal language  
(US English) will be used, otherwise,  
the language to be used may be speci-  
An exact match for fied in the format language-region.  
one of the folder  
names under the  
SoundPointIPLo-  
calization folder on  
the boot server.  
lcl.ml.lang.menu.x  
String in the format Multiple lcl.ml.lang.menu.x attributes  
language_region  
are supported - as many languages as  
are desired. However, the  
lcl.ml.lang.menu.x attributes must be  
sequential (lcl.ml.lang.menu.1,  
lcl.ml.lang.menu.2,  
lcl.ml.lang.menu.3, ...,  
lcl.ml.lang.menu.N) with no gaps and  
the strings must exactly match a folder  
name under the SoundPointIPLocaliza-  
tion folder on the boot server for the  
phone to be able to locate the dictio-  
nary file.  
lcl.ml.lang.clock.x.24HourClock  
lcl.ml.lang.clock.x.format  
0,1  
If attribute present, overrides  
lcl.datetime.time.24HourClock;  
If 1, display time in 24-hour clock  
mode rather than am/pm.  
string which  
If attribute present, overrides  
lcl.datetime.date.format;  
D = day of week  
d = day  
includes ‘D’, ‘d’  
and ‘M’ and two  
optional commas  
M = month  
Up to two commas may be included.  
For example: D,dM = Thursday, 3 July  
or Md,D = July 3, Thursday  
The field may contain 0, 1 or 2 com-  
mas which can occur only between  
characters and only one at a time. For  
example: “D,,dM” is illegal.  
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Permitted  
Values  
Attribute  
Interpretation  
lcl.ml.lang.clock.x.longFormat  
0, 1  
If attribute present, overrides  
lcl.datetime.date.longFormat;  
If 1, display the day and month in long  
format (Friday/November), otherwise  
use abbreviations (Fri/Nov).  
lcl.ml.lang.clock.x.dateTop  
lcl.ml.lang.y.list  
0, 1  
If attribute present, overrides  
lcl.datetime.date.dateTop;  
If 1, display date above time, otherwise  
display time above date.  
“All” or a comma-  
separated list  
A list of the languages supported on  
hardware platform ‘y’ where ‘y’ can be  
IP_500 or IP_600.  
4.6.1.3.1.1 Adding New Languages  
To add new languages to those included with the distribution:  
1. Create a new dictionary file based on an existing one.  
2. Change the strings making sure to encode the XML file in UTF-8 but also  
ensuring the UTF-8 characters chosen are within the Unicode character  
ranges indicated in 3.5.1 Multilingual User Interface on page 61.  
3. Place the file in an appropriately named folder according to the format  
language_region parallel to the other dictionary files under the SoundPoint-  
IPLocalization folder on the boot server.  
4. Add a lcl.ml.lang.clock.menu.x attribute to the configuration file.  
5. Add lcl.ml.lang.clock.x.24HourClock, lcl.ml.lang.clock.x.format,  
lcl.ml.lang.clock.x.longFormat and lcl.ml.lang.clock.x.dateTop attributes  
and set them according to the regional preferences.  
6. (Optional) Set lcl.ml.lang to be the new language_region string.  
4.6.1.3.2 Date and Time <datetime/>  
Permitted  
Values  
Attribute  
Interpretation  
lcl.datetime.time.24HourClock  
0,1  
If 1, display time in 24-hour clock mode rather  
than a.m./p.m.  
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Permitted  
Values  
Attribute  
Interpretation  
lcl.datetime.date.format  
string which  
includes ‘D’,  
‘d’ and ‘M’  
and two  
Controls format of date string.  
D = day of week  
d = day  
M = month  
optional com-  
mas  
Up to two commas may be included.  
For example: D,dM = Thursday, 3 July or  
Md,D = July 3, Thursday  
The field may contain 0, 1 or 2 commas which  
can occur only between characters and only  
one at a time. For example: “D,,dM” is illegal.  
lcl.datetime.date.longFormat  
lcl.datetime.date.dateTop  
0,1  
If 1, display the day and month in long format  
(Friday/November), otherwise, use abbrevia-  
tions (Fri/Nov).  
0, 1  
If 1, display date above time else display time  
above date.  
4.6.1.4 User Preferences <user_preferences/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
up.headsetMode  
0,1  
0
If set to 1, the headset will be selected as  
the preferred transducer after its first use  
until the headset key is pressed again;  
otherwise, hands-free will be selected  
preferentially over the headset.  
up.useDirectoryNames  
0,1  
0
0
If set to 1, the name fields of directory  
entries which match incoming calls will  
be used for caller identification display  
and in the call lists instead of the name  
provided through network signaling.  
up.oneTouchVoiceMail  
0, 1  
If set to 1, the voice mail summary dis-  
play is bypassed and voice mail is dialed  
directly (if configured).  
up.welcomeSoundEnabled  
0, 1  
0, 1  
1
0
If set to 1, play welcome sound effect  
after a reboot.  
up.welcomeSoundOnWarm-  
BootEnabled  
If set to 1, play welcome sound effect on  
warm as well as cold boots, otherwise  
only a cold boot will trigger the wel-  
come sound effect.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
up.localClockEnabled  
0, 1  
1
If set to 1, display the date and time on  
the idle display  
4.6.1.5 Tones <tones/>  
This section describes configuration items for the tone resources available in the  
phone.  
4.6.1.5.1 Dual Tone Multi-Frequency <DTMF/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
tone.dtmf.level  
-33 to -3  
-15  
Level of the high frequency compo-  
nent of the DTMF digit measured in  
dBm0; the low frequency tone will  
be two dB lower.  
tone.dtmf.onTime  
positive  
integer  
50  
When a sequence of DTMF tones is  
played out automatically, this is the  
length of time in milliseconds the  
tones will be generated for; this is  
also the minimum time the tone will  
be played for when dialing manually  
(even if key press is shorter).  
tone.dtmf.offTime  
positive  
integer  
50  
When a sequence of DTMF tones is  
played out automatically, this is the  
length of time in milliseconds the  
phone will pause between digits;  
this is also the minimum inter-digit  
time when dialing manually.  
tone.dtmf.chassis.masking  
0, 1  
0
If set to 1, DTMF tones will be sub-  
stituted with a non-DTMF pacifier  
tone when dialing in hands-free  
mode. This prevents DTMF digits  
being broadcast to other surrounding  
telephony devices or being inadvert-  
ently transmitted in-band due to  
local acoustic echo.  
Note: tone.dtmf.chassis.masking  
should only be enabled when  
tone.dtmf.viaRtp is disabled.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
tone.dtmf.stim.pac.offHookOnly  
tone.dtmf.viaRtp  
0, 1  
0
1
Not currently used.  
0, 1  
If set to 1, encode DTMF in the  
active RTP stream, otherwise,  
DTMF may be encoded within the  
signaling protocol only when the  
protocol offers the option.  
Note: tone.dtmf.chassis.masking  
should be enabled when  
tone.dtmf.viaRtp is disabled.  
tone.dtmf.rfc2833Control  
0, 1  
1
If set to 1, the phone will indicate a  
preference for encoding DTMF  
through RFC 2833 format in its Ses-  
sion Description Protocol (SDP)  
offers by showing support for the  
phone-event payload type; this does  
not affect SDP answers, these will  
always honor the DTMF format  
present in the offer since the phone  
has native support for RFC 2833.  
tone.dtmf.rfc2833Payload  
96-127  
101  
The phone-event payload encoding  
in the dynamic range to be used in  
SDP offers.  
4.6.1.5.2 Chord Sets <chord_sets/>  
Chord sets are the building blocks of sound effects that use synthesized rather than  
sampled audio (most call progress and ringer sound effects). A chord-set is a multi-fre-  
quency note with an optional on/off cadence. A chord-set can contain up to four fre-  
quency components generated simultaneously, each with its own level.  
There are three blocks of chord sets:  
• callProg (used for call progress sound effect patterns)  
• ringer  
• misc (miscellaneous)  
All three blocks use the same chord set specification format.  
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In the following table, x is the chord-set number and cat is one of callProg, ringer, or  
misc.  
Permitted  
Values  
Attribute  
Interpretation  
tone.chord.cat.x.freq.y  
0-1600  
Frequency for this component in Hertz; up to four  
chord-set components can be specified (y=1, 2, 3,  
4).  
tone.chord.cat.x.level.y  
-57 to 3  
Level of this component in dBm0.  
tone.chord.cat.x.onDur  
positive  
integer  
On duration in milliseconds, 0=infinite.  
tone.chord.cat.x.offDur  
positive  
integer  
Off duration in milliseconds, 0=infinite.  
tone.chord.cat.x.repeat  
positive  
integer  
Specifies how many times the ON/OFF cadence  
is repeated, 0=infinite.  
4.6.1.6 Sampled Audio for Sound Effects <sampled_audio/>  
The following sampled audio WAVE file (.wav) formats are supported:  
• mono 8 kHz G.711 μ-Law  
• G.711 A-Law  
• L16/160006 (16-bit, 16 kHz sampling rate, mono)  
The phone uses built-in wave files for some sound effects. The built-in wave files can  
be replaced with files downloaded from the boot server or from the Internet, however,  
these are stored in volatile memory so the files will need to remain accessible should  
the phone need to be rebooted. Files will be truncated to a maximum size of 300 kilo-  
bytes.  
®
®
6. L16/16000 is not supported on SoundPoint IP 300, 301 and SoundStation IP 4000 phones.  
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In the following table, x is the sampled audio file number.  
Attribute  
Permitted Values  
Interpretation  
saf.x  
Null OR valid path  
name OR an RFC  
1738-compliant URL  
to a HTTP, FTP, or  
TFTP wave file  
resource.  
If Null, the phone will use a built-in file.  
If set to a path name, the phone will attempt to download  
this file at boot time from the boot server.  
If set to a URL, the phone will attempt to download this  
file at boot time from the Internet.  
Note: Refer to the  
above wave file for-  
mat restrictions.  
Note: A TFTP URL is expected to be in the format: tftp:/  
/<host>/[pathname]<filename>, for example: tftp://  
somehost.example.com/sounds/example.wav  
The following table defines the default usage of the sampled audio files with the  
phone:  
Sampled Audio File  
Default use within phone (pattern reference)  
Welcome Sound Effect (se.pat.misc.7)  
Ringer 13 (se.pat.ringer.13)  
Ringer 14 (se.pat.ringer.14)  
Ringer 15 (se.pat.ringer.15)  
Ringer 16 (se.pat.ringer.16)  
Ringer 17 (se.pat.ringer.17)  
Ringer 18 (se.pat.ringer.18)  
Ringer 19 (se.pat.ringer.19)  
Ringer 20 (se.pat.ringer.20)  
Ringer 21 (se.pat.ringer.21)  
Ringer 22 (se.pat.ringer.22)  
Not used.  
1
2
3
4
5
6
7
8
9
10  
11  
12-24  
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4.6.1.7 Sound Effects <sound_effects/>  
The phone uses both synthesized (based on the chord-sets described earlier) and sam-  
pled audio sound effects. Sound effects are defined by patterns: rudimentary  
sequences of chord-sets, silence periods, and wave files.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
se.stutterOnVoiceMail  
0, 1  
1
If set to 1, stuttered dial tone is used in place  
of normal dial tone to indicate that one or  
more messages (voice mail) are waiting at  
the message center.  
se.appLocalEnabled  
0, 1  
1
If set to 1, local user interface sound effects  
such as confirmation/error tones, will be  
enabled.  
4.6.1.7.1 Patterns <patterns/>  
Patterns use a simple script language that allows different chord sets or wave files to  
be strung together with periods of silence. The script language uses the following  
instructions:  
Instruction  
Meaning  
Example  
sampled (n)  
Play sampled audio  
se.pat.callProg.x.inst.y.type =”sampled” (sampled audio  
file instruction type)  
a
file n  
se.pat.callProg.x.inst.y.value =”3” (specifies sampled  
audio file 3)  
chord (n, d)  
Play chord set n (d is se.pat.callProg.x.inst.y.type = “chord” (chord set  
optional and allows  
the chord set ON  
duration to be over-  
ridden to d millisec-  
onds)  
instruction type)  
se.pat.callProg.x.inst.y.value = “3” (specifies call  
progress chord set 3)  
se.pat.callProg.x.inst.y.param = “2000” (override ON  
duration of chord set to 2000 milliseconds)  
silence (d)  
Play silence for d  
milliseconds (Rx  
audio is not muted)  
se.pat.callProg.x.inst.y.type = “silence” (silence instruc-  
tion type)  
se.pat.callProg.x.inst.y.value = “300” (specifies silence  
is to last 300 milliseconds)  
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Instruction  
Meaning  
Example  
branch (n)  
Advance n instruc-  
tions and execute  
that instruction (n  
must be negative and  
must not branch  
beyond the first  
instruction)  
se.pat.callProg.x.inst.y.type = “branch” (branch instruc-  
tion type)  
se.pat.callProg.x.inst.y.value = “-5” (step back 5 instruc-  
tions and execute that instruction)  
a. Currently, patterns that use the sampled instruction are limited to the following format:  
sampled followed by optional silence and optional branch back to the beginning.  
In the following table, x is the pattern number, y is the instruction number. Both x and  
y need to be sequential. There are three categories of sound effect patterns: callProg  
(call progress patterns), ringer and misc (miscellaneous).  
Permitted  
Values  
Attribute  
Interpretation  
se.pat.callProg.x.name  
UTF-8  
encoded  
string  
Used for identification purposes in the user interface (cur-  
rently used for ringer patterns only); for patterns that use  
a sampled audio file which has been overridden by a  
downloaded replacement, the se.pat.ringer.x.name  
parameter will be overridden in the user interface by the  
file names of the wave file.  
se.pat.call-  
sampled  
As above.  
Prog.x.inst.y.type  
OR chord  
OR silence  
OR branch  
se.pat.call-  
Prog.x.inst.y.value  
integer  
Instruction type: Interpretation:  
sampled sampled audio file number  
chord chord set number  
silence silence duration in ms  
branch number of instructions to advance  
se.pat.call-  
Prog.x.inst.y.param  
positive  
integer  
If instruction type is chord, this optional parameter speci-  
fies the on duration to be used, overriding the on duration  
specified in the chord-set definition.  
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4.6.1.7.1.1 Call Progress Patterns  
The following table maps call progress patterns to their usage within the phone.  
Call progress  
pattern number  
Use within phone  
dial tone  
1
2
busy tone  
3
ring back tone  
4
reorder tone  
5
stuttered dial tone  
call waiting tone  
alternate call waiting tone (distinctive)  
confirmation tone  
howler tone (off-hook warning)  
record warning  
6
7
8
9
10  
11  
12  
13  
14  
15  
message waiting tone  
alerting  
intercom announcement tone  
barge-in tone  
secondary dial tone  
4.6.1.7.1.2 Ringer Patterns  
The following table maps ringer pattern numbers to their default descriptions.  
Ringer pattern  
number  
Default description  
a
1
Silent Ring  
2
3
4
5
6
7
Low Trill  
Low Double Trill  
Medium Trill  
Medium Double Trill  
High Trill  
High Double Trill  
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Ringer pattern  
number  
Default description  
Highest Trill  
Highest Double Trill  
Beeble  
8
9
10  
11  
12  
13  
Triplet  
Ringback-style  
b
Sampled audio file 2  
Sampled audio file 3  
Sampled audio file 4  
Sampled audio file 5  
Sampled audio file 6  
Sampled audio file 7  
Sampled audio file 8  
Sampled audio file 9  
14  
15  
16  
17  
18  
19  
20  
21  
22  
Sampled audio file 10  
Sampled audio file 11  
a. Silent Ring will only provide a visual indication of an incoming  
call, but no audio indication.  
b. Sampled audio files 1-21 all use the same built-in file unless that  
file has been replaced with a downloaded file. For more informa-  
tion, refer to 4.6.1.6 Sampled Audio for Sound Effects  
<sampled_audio/> on page 101.  
4.6.1.7.1.3 Miscellaneous Patterns  
The following table maps miscellaneous patterns to their usage within the phone.  
Miscellaneous  
pattern number  
Use within phone  
1
2
3
4
5
6
new message waiting indication  
new instant message  
Not used.  
local hold notification  
positive confirmation  
negative confirmation  
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Miscellaneous  
pattern number  
Use within phone  
7
welcome (boot up)  
4.6.1.7.2 Ring type <ringType/>  
Ring type is used to define a simple class of ring to be applied based on some creden-  
tials that are usually carried within the network protocol. The ring class includes  
attributes such as call-waiting and ringer index, if appropriate. The ring class can use  
one of four types of ring that are defined as follows:  
ring  
Play a specified ring pattern or call waiting indication.  
visual  
Provide only a visual indication (no audio indication) of incoming call (no  
ringer needs to be specified).  
a
answer  
Provide auto-answer on incoming call .  
a
ring-answer  
Provide auto answer on incoming call after a ring period .  
a. Note that auto-answer on incoming call is currently only applied if there is no other  
call in progress on the phone at the time.  
In the following table, x is the ring class number. The x index needs to be sequential.  
Attribute  
Permitted Values  
Interpretation  
se.rt.enabled  
0,1  
Set to 1 to enable the ring type feature within  
the phone, 0 otherwise.  
se.rt.modification.enabled 0,1  
Set to 1 to allow user modification through  
local user interface of the pre-defined ring  
a
type enabled for modification .  
se.rt.x.name  
se.rt.x.type  
UTF-8 encoded string  
Used for identification purposes in the user  
a
interface .  
ring OR visual OR  
answer OR ring-  
answer  
As defined in table above.  
se.rt.x.ringer  
integer - only relevant  
if the type is set to  
The ringer index to be used for this class of  
ring. The ringer index should match one of  
‘ring’ or ‘ring-answer’ 4.6.1.7.1.2 Ringer Patterns on page 105.  
se.rt.x.callWait  
integer - only relevant  
if the type is set to  
The call waiting index to be used for this  
class of ring. The call waiting index should  
‘ring’ or ‘ring-answer’ match one defined in 4.6.1.7.1.1 Call  
Progress Patterns on page 105.  
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Attribute  
Permitted Values  
Interpretation  
se.rt.x.timeout  
positive integer - only  
relevant if the type is  
set to ‘ring-answer’.  
Default value is 2000.  
The duration of the ring in milliseconds  
before the call is auto answered. If this field  
is omitted or is left blank, a value of 2000 is  
used.  
se.rt.x.mod  
0,1  
Set to 1 if the user interface should allow for  
modification by the user of the ringer index  
used for this ring class.  
a. Modification through user interface will be implemented in a future release.  
4.6.1.8 Voice Settings <voice/>  
4.6.1.8.1 Voice Coding Algorithms <codecs/>  
The following voice codecs are supported:  
MIME  
Type  
Sample  
Bit Rate Rate  
Effective Audio  
Bandwidth  
Algorithm  
G.711μ-law  
G.711a-law  
G.729AB  
Label  
Frame Size  
PMCU  
PCMA  
G729  
G711mu  
G711A  
64 Kbps 8 Ksps  
64 Kbps 8 Ksps  
10ms - 80ms  
10ms - 80ms  
10ms - 80ms  
3.5KHz  
3.5KHz  
3.5KHz  
G729AB 8 Kbps  
8 Ksps  
4.6.1.8.1.1 Codec Preferences <preferences/>  
Permitted  
Values  
Attribute  
Default Interpretation  
voice.codecPref.G711Mu  
voice.codecPref.G711A  
voice.codecPref.G729AB  
Null, 1-3  
1
2
3
Specifies the codec preferences for  
SoundPoint® IP 430, 500, 501, 600  
and 601 platforms.  
1 = highest  
3 = lowest  
Null = do not use  
Give each codec a unique priority,  
this will dictate the order used in  
SDP negotiations.  
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Permitted  
Values  
Attribute  
Default Interpretation  
voice.codecPref.IP_300.G711Mu  
voice.codecPref.IP_300.G711A  
voice.codecPref.IP_300.G729AB  
voice.codecPref.IP_4000.G711Mu  
voice.codecPref.IP_4000.G711A  
Null, 1-3  
1
2
3
1
2
Specifies the codec preferences for  
SoundPoint® IP 300 and 301 plat-  
forms. Interpretation as above.  
Null, 1-3  
Specifies the codec preferences for  
the SoundStation® IP 4000 platform.  
Interpretation as above.  
voice.codecPref.IP_4000.G729AB  
Null  
Not supported by default so that  
G.711Mu and G.711A local confer-  
ences can be supported. This restric-  
tion will be removed in a future  
release.  
4.6.1.8.1.2 Codec Profiles <profiles/>  
The following profile attributes can be adjusted for each of the three supported codecs.  
In the table, x=G711Mu, G711A, or G729AB.  
Permitted  
Values  
Attribute  
Interpretation  
voice.audioProfile.x.payloadSize  
10, 20, 30,  
...80  
Preferred Tx payload size in millisec-  
onds to be provided in SDP offers and  
used in the absence of ptime negotia-  
tions. This is also the range of supported  
Rx payload sizes.  
voice.audioProfile.x.jitterBufferMin  
20, 40, 50,  
The smallest jitter buffer depth (in milli-  
60, ... (multi- seconds) that must be achieved before  
ple of 10)  
play out begins for the first time. Once  
this depth has been achieved initially, the  
depth may fall below this point and play  
out will still continue. This parameter  
should be set to the smallest possible  
value which is at least two packet pay-  
loads, and larger than the expected short  
term average jitter. The IP4000 values  
are the same as the IP30x values.  
voice.audioProfile.x.jitterBufferShrink  
10, 20, 30, ... The absolute minimum duration time (in  
(multiple of  
10)  
milliseconds) of RTP packet Rx with no  
packet loss between jitter buffer size  
shrinks. Use smaller values (1000 ms) to  
minimize the delay on known good net-  
works. Use larger values to minimize  
packet loss on networks with large jitter  
(3000 ms).  
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Permitted  
Values  
Attribute  
Interpretation  
voice.audioProfile.x.jitterBufferMax  
> jitterBuf-  
ferMin,  
multiple of  
10,  
<=500 for IP  
430, 500,  
The largest jitter buffer depth to be sup-  
ported (in milliseconds). Jitter above this  
size will always cause lost packets. This  
parameter should be set to the smallest  
possible value that will support the  
expected network jitter.  
501, and 600,  
<= 160 for IP  
300 and 301  
4.6.1.8.2 Volume Persistence <volume/>  
The user’s selection of the receive volume during a call can be remembered between  
calls. This can be configured per termination (handset, headset and hands-free/chas-  
sis). In some countries regulations exist which dictate that receive volume should be  
reset to nominal at the start of each call on handset and headset.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
voice.volume.persist.handset  
voice.volume.persist.headset  
voice.volume.persist.handsfree  
0, 1  
0
0
1
If set to 1, the receive volume will be  
remembered between calls.  
0, 1  
If set to 0, the receive volume will be  
reset to nominal at the start of each  
call.  
0, 1  
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4.6.1.8.3 Gains <gains/>  
The default gain settings have been carefully adjusted to comply with the TIA-810-A  
digital telephony standard.  
Note  
Polycom recommends that you do not change these values.  
Attribute  
Default  
0
voice.gain.rx.analog.handset  
voice.gain.rx.analog.headset  
0
voice.gain.rx.analog.chassis  
0
voice.gain.rx.analog.chassis.IP_300  
voice.gain.rx.analog.chassis.IP_430  
voice.gain.rx.analog.chassis.IP_4000  
voice.gain.rx.analog.chassis.IP_601  
voice.gain.rx.analog.ringer  
-6  
0
3
6
0
voice.gain.rx.analog.ringer.IP_300  
voice.gain.rx.analog.ringer.IP_430  
voice.gain.rx.analog.ringer.IP_4000  
voice.gain.rx.analog.ringer.IP_601  
voice.gain.rx.digital.handset  
-6  
0
3
6
-15  
-21  
0
voice.gain.rx.digital.headset  
voice.gain.rx.digital.chassis  
voice.gain.rx.digital.chassis.IP_430  
voice.gain.rx.digital.chassis.IP_4000  
voice.gain.rx.digital.chassis.IP_601  
voice.gain.rx.digital.ringer  
0
0
0
-21  
-21  
-21  
-21  
-14  
-24  
voice.gain.rx.digital.ringer.IP_430  
voice.gain.rx.digital.ringer.IP_4000  
voice.gain.rx.digital.ringer.IP_601  
voice.gain.rx.analog.handset.sidetone  
voice.gain.rx.analog.headset.sidetone  
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Attribute  
Default  
voice.gain.tx.analog.handset  
12  
3
voice.gain.tx.analog.headset  
voice.gain.tx.analog.chassis  
3
voice.gain.tx.analog.chassis.IP_300  
voice.gain.tx.analog.chassis.IP_430  
voice.gain.tx.analog.chassis.IP_4000  
voice.gain.tx.analog.chassis.IP_601  
voice.gain.tx.digital.handset  
0
42  
3
0
0
voice.gain.tx.digital.headset  
0
voice.gain.tx.digital.chassis  
3
voice.gain.tx.digital.chassis.IP_4000  
voice.gain.tx.digital.chassis.IP_601  
voice.gain.tx.digital.chassis.IP_430  
voice.gain.tx.analog.preamp.handset  
voice.gain.tx.analog.preamp.headset  
voice.gain.tx.analog.preamp.chassis  
0
6
0
14  
23  
32  
voice.gain.tx.analog.preamp.chassis.IP_430 32  
voice.gain.tx.analog.preamp.chassis.IP_601 32  
voice.handset.rxag.adjust.IP_430  
voice.handset.txag.adjust.IP_430  
voice.handset.sidetone.adjust.IP_430  
voice.headset.rxag.adjust.IP_430  
voice.headset.txag.adjust.IP_430  
voice.headset.sidetone.adjust.IP_430  
1
21  
-12  
1
39  
-3  
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4.6.1.8.4 Acoustic Echo Cancellation <AEC/>  
These settings control the performance of the speakerphone acoustic echo canceller.  
Note  
Polycom recommends that you do not change these values.  
Attribute  
Default  
0
voice.aec.hs.enable  
voice.aec.hs.lowFreqCutOff  
voice.aec.hs.highFreqCutOff  
voice.aec.hs.erlTab_0_300  
voice.aec.hs.erlTab_300_600  
voice.aec.hs.erlTab_600_1500  
voice.aec.hs.erlTab_1500_3500  
voice.aec.hs.erlTab_3500_7000  
voice.aec.hd.enable  
100  
7000  
-24  
-24  
-24  
-24  
-24  
0
voice.aec.hd.lowFreqCutOff  
voice.aec.hd.highFreqCutOff  
voice.aec.hd.erlTab_0_300  
voice.aec.hd.erlTab_300_600  
voice.aec.hd.erlTab_600_1500  
voice.aec.hd.erlTab_1500_3500  
voice.aec.hd.erlTab_3500_7000  
voice.aec.hf.enable  
100  
7000  
-24  
-24  
-24  
-24  
-24  
1
voice.aec.hf.lowFreqCutOff  
voice.aec.hf.highFreqCutOff  
voice.aec.hf.erlTab_0_300  
voice.aec.hf.erlTab_300_600  
voice.aec.hf.erlTab_600_1500  
voice.aec.hf.erlTab_1500_3500  
voice.aec.hf.erlTab_3500_7000  
100  
7000  
-6  
-6  
-6  
-6  
-6  
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4.6.1.8.5 Acoustic Echo Suppression <AES/>  
These settings control the performance of the speakerphone acoustic echo suppressor.  
Note  
Polycom recommends that you do not change these values.  
Attribute  
Default  
voice.aes.hs.enable  
0
7
0
0
1
7
7
6
6
5
4
4
3
2
10  
9
8
7
6
5
4
3
2
voice.aes.hs.duplexBalance  
voice.aes.hd.enable  
voice.aes.hd.duplexBalance  
voice.aes.hf.enable  
voice.aes.hf.duplexBalance.0  
voice.aes.hf.duplexBalance.1  
voice.aes.hf.duplexBalance.2  
voice.aes.hf.duplexBalance.3  
voice.aes.hf.duplexBalance.4  
voice.aes.hf.duplexBalance.5  
voice.aes.hf.duplexBalance.6  
voice.aes.hf.duplexBalance.7  
voice.aes.hf.duplexBalance.8  
voice.aes.hf.duplexBalance.IP_4000.0  
voice.aes.hf.duplexBalance.IP_4000.1  
voice.aes.hf.duplexBalance.IP_4000.2  
voice.aes.hf.duplexBalance.IP_4000.3  
voice.aes.hf.duplexBalance.IP_4000.4  
voice.aes.hf.duplexBalance.IP_4000.5  
voice.aes.hf.duplexBalance.IP_4000.6  
voice.aes.hf.duplexBalance.IP_4000.7  
voice.aes.hf.duplexBalance.IP_4000.8  
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4.6.1.8.6 Background Noise Suppression <NS/>  
These settings control the performance of the transmit background noise suppression  
feature.  
Note  
Polycom recommends that you do not change these values.  
Attribute  
Default  
voice.ns.hs.enable  
0
voice.ns.hs.signalAttn  
voice.ns.hs.silenceAttn  
voice.ns.hd.enable  
-6  
-9  
0
voice.ns.hd.signalAttn  
voice.ns.hd.silenceAttn  
voice.ns.hf.enable  
0
0
1
voice.ns.hf.signalAttn  
voice.ns.hf.silenceAttn  
voice.ns.hf.IP_4000.enable  
voice.ns.hf.IP_4000.signalAttn  
-6  
-9  
1
-6  
voice.ns.hf.IP_4000.silenceAttn -9  
4.6.1.8.7 Automatic Gain Control <AGC/>  
7
These settings control the performance of the transmit automatic gain control feature.  
Note  
Polycom recommends that you do not change these values.  
Attribute  
Default  
voice.agc.hs.enable  
voice.agc.hd.enable  
voice.agc.hf.enable  
0
0
0
7. Automatic Gain Control will be implemented in a future release.  
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4.6.1.8.8 Receive Equalization <RXEQ/>  
These settings control the performance of the receive equalizaton feature.  
Note  
Polycom recommends that you do not change these values.  
Attribute  
Default  
voice.rxEq.hs.IP_430.preFilter.enable  
voice.rxEq.hs.IP_500.preFilter.enable  
voice.rxEq.hs.IP_600.preFilter.enable  
voice.rxEq.hs.IP_601.preFilter.enable  
voice.rxEq.hs.IP_430.postFilter.enable  
voice.rxEq.hs.IP_500.postFilter.enable  
voice.rxEq.hs.IP_600.postFilter.enable  
voice.rxEq.hs.IP_601.postFilter.enable  
voice.rxEq.hd.IP_430.preFilter.enable  
voice.rxEq.hd.IP_500.preFilter.enable  
voice.rxEq.hd.IP_600.preFilter.enable  
voice.rxEq.hd.IP_601.preFilter.enable  
voice.rxEq.hd.IP_430.postFilter.enable  
voice.rxEq.hd.IP_500.postFilter.enable  
voice.rxEq.hd.IP_600.postFilter.enable  
voice.rxEq.hd.IP_601.postFilter.enable  
voice.rxEq.hf.IP_430.preFilter.enable  
voice.rxEq.hf.IP_500.preFilter.enable  
voice.rxEq.hf.IP_600.preFilter.enable  
voice.rxEq.hf.IP_601.preFilter.enable  
voice.rxEq.hf.IP_4000.preFilter.enable  
voice.rxEq.hf.IP_430.postFilter.enable  
voice.rxEq.hf.IP_500.postFilter.enable  
voice.rxEq.hf.IP_600.postFilter.enable  
voice.rxEq.hf.IP_601.postFilter.enable  
1
1
1
1
0
0
0
0
0
0
0
0
0
0
0
0
1
1
1
1
0
0
1
1
1
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Attribute  
Default  
voice.rxEq.hf.IP_4000.postFilter.enable  
0
4.6.1.8.9 Transmit Equalization <TXEQ/>  
These settings control the performance of the hands-free transmit equalization feature.  
Note  
Polycom recommends that you do not change these values.  
Attribute  
Default  
voice.txEq.hs.IP_430.preFilter.enable  
voice.txEq.hs.IP_500.preFilter.enable  
voice.txEq.hs.IP_600.preFilter.enable  
voice.txEq.hs.IP_601.preFilter.enable  
voice.txEq.hs.IP_430.postFilter.enable  
voice.txEq.hs.IP_500.postFilter.enable  
voice.txEq.hs.IP_600.postFilter.enable  
voice.txEq.hs.IP_601.postFilter.enable  
voice.txEq.hd.IP_430.preFilter.enable  
voice.txEq.hd.IP_500.preFilter.enable  
voice.txEq.hd.IP_600.preFilter.enable  
voice.txEq.hd.IP_601.preFilter.enable  
voice.txEq.hd.IP_430.postFilter.enable  
voice.txEq.hd.IP_500.postFilter.enable  
voice.txEq.hd.IP_600.postFilter.enable  
voice.txEq.hd.IP_601.postFilter.enable  
voice.txEq.hf.IP_430.preFilter.enable  
voice.txEq.hf.IP_500.preFilter.enable  
voice.txEq.hf.IP_600.preFilter.enable  
voice.txEq.hf.IP_601.preFilter.enable  
voice.txEq.hf.IP_4000.preFilter.enable  
0
0
0
0
1
1
1
1
0
0
0
0
0
0
0
0
0
0
0
0
0
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Attribute  
Default  
voice.txEq.hf.IP_430.postFilter.enable  
voice.txEq.hf.IP_500.postFilter.enable  
voice.txEq.hf.IP_600.postFilter.enable  
voice.txEq.hf.IP_601.postFilter.enable  
voice.txEq.hf.IP_4000.postFilter.enable  
1
1
1
1
0
4.6.1.8.10 Voice Activity Detection <VAD/>  
These settings control the performance of the voice activity detection (silence suppres-  
sion) feature.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
voice.vadEnable  
voice.vadThresh  
0, 1  
0
If set to 1, enable VAD.  
integer from 15  
0 to 30  
The threshold for determining what is active  
voice and what is background noise in dB. This  
does not apply to G.729AB codec operation  
which has its own built-in VAD function.  
4.6.1.9 Quality of Service <QOS/>  
These settings control the Quality of Service (QOS) options.  
4.6.1.9.1 Ethernet IEEE 802.1p/Q <Ethernet/>  
These settings control the 802.1p/Q user_priority field.  
4.6.1.9.1.1 RTP <RTP/>  
These parameters apply to RTP packets.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ethernet.rtp.user_priority 0-7  
5
User-priority used for RTP packets.  
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4.6.1.9.1.2 Call Control <CallControl/>  
These parameters apply to call control packets, such as the network protocol signaling.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ethernet.callControl.user_priority  
0-7  
5
User-priority used for call con-  
trol packets.  
4.6.1.9.1.3 Other <Other/>  
These default parameter values are used for all packets which are not set explicitly.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ethernet.other.user_priority  
0-7  
2
User-priority used for packets that  
do not have a per-protocol setting.  
4.6.1.9.2 IP TOS <IP/>  
These settings control the “type of service” field in outgoing packets.  
4.6.1.9.2.1 RTP <RTP/>  
These parameters apply to RTP packets.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ip.rtp.dscp  
0 to 63 or  
EF or  
Null  
The differentiated services  
codepoints of packets.  
any of  
If set to null, the values below  
of min_delay,  
max_throughput,  
max_reliability, min_cost, and  
precedence are used. Other-  
wise, these values are overrid-  
den.  
AF11,AF12,  
AF13,AF21,  
AF22,AF23,  
AF31,AF32,  
AF33,AF41,  
AF42,AF43  
qos.ip.rtp.min_delay  
0, 1  
1
If set to 1, set min-delay bit in  
the IP TOS field of the IP  
header, or else don’t set it.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ip.rtp.max_throughput  
0, 1  
1
If set to 1, set max-throughput  
bit in the IP TOS field of the IP  
header, or else don’t set it.  
qos.ip.rtp.max_reliability  
qos.ip.rtp.min_cost  
0, 1  
0, 1  
0-7  
0
0
5
If set to 1, set max-reliability  
bit in the IP TOS field of the IP  
header, or else don’t set it.  
If set to 1, set min-cost bit in  
the IP TOS field of the IP  
header, or else don’t set it.  
qos.ip.rtp.precedence  
If set to 1, set precedence bits  
in the IP TOS field of the IP  
header, or else don’t set them.  
4.6.1.9.2.2 Call Control <CallControl/>  
These parameters apply to call control packets, such as the network protocol signaling.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ip.callControl.dscp  
0 to 63 or  
EF or  
Null  
The differentiated services  
codepoints of packets.  
any of  
If set to null, the values below  
of min_delay, max_throughput,  
max_reliability, min_cost, and  
precedence are used. Other-  
wise, these values are overrid-  
den.  
AF11,AF12,  
AF13,AF21,  
A F 2 2 , A F 2 3 ,  
AF31,AF32,  
AF33,AF41,  
AF42,AF43  
qos.ip.callControl.min_delay  
qos.ip.callControl.max_throughput  
qos.ip.callControl.max_reliability  
qos.ip.callControl.min_cost  
0, 1  
0, 1  
0, 1  
0, 1  
1
0
0
0
If set to 1, set min-delay bit in  
the IP TOS field of the IP  
header, or else don’t set it.  
If set to 1, set max-throughput  
bit in the IP TOS field of the IP  
header, or else don’t set it.  
If set to 1, set max-reliability  
bit in the IP TOS field of the IP  
header, or else don’t set it.  
If set to 1, set min-cost bit in  
the IP TOS field of the IP  
header, or else don’t set it.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ip.callControl.precedence  
0-7  
5
If set to 1, set precedence bits  
in the IP TOS field of the IP  
header, or else don’t set them.  
4.6.1.10 Basic TCP/IP <TCP_IP/>  
4.6.1.10.1 Network Monitoring <netMon/>  
Note  
Polycom recommends that you do not change these values.  
Attribute  
Permitted Values  
0, 1  
Default  
tcpIpApp.netMon.enabled  
tcpIpApp.netMon.period  
1
1 to 86400  
30  
4.6.1.10.2 Time Synchronization <SNTP/>  
The following table describes the parameters used to set up time synchronization and  
daylight savings time. The defaults shown will enable daylight savings time (DST) for  
North America.  
Daylight savings defaults:  
• Do not use fixed day, use first or last day of week in the month.  
• Start DST on the first Sunday in April at 2 am.  
• Stop DST on the last Sunday in October at 2 am.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
tcpIpApp.sntp.resyncPeriod  
positive  
integer  
86400  
(24  
hours)  
Time in seconds  
between Simple Net-  
work Time Protocol  
(SNTP) re-syncs.  
tcpIpApp.sntp.address  
valid host  
name or IP  
address  
clock  
Address of the SNTP  
server.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
tcpIpApp.sntp.address.overrideDHCP  
0, 1  
0
These parameters deter-  
mine whether configu-  
ration file parameters  
override DHCP parame-  
ters for the SNTP server  
address and Greenwich  
Mean Time (GMT) off-  
set. If set to 0, DHCP  
values will override  
configuration file  
parameters. If set to 1,  
the configuration file  
parameters will override  
DHCP values.  
tcpIpApp.sntp.gmtOffset  
positive or  
negative  
integer  
-28800  
(Pacific  
time)  
Offset in seconds of the  
local time zone from  
GMT.  
Note: 3600 seconds per  
hour  
tcpIpApp.sntp.gmtOffset.overrideDHCP  
0, 1  
0
These parameters deter-  
mine whether configu-  
ration file parameters  
override DHCP parame-  
ters for the SNTP server  
address and GMT off-  
set. If set to 0, DHCP  
values will override  
configuration file  
parameters. If set to 1,  
the configuration file  
parameters will override  
DHCP values.  
tcpIpApp.sntp.daylightSavings.enable  
0, 1  
0, 1  
1
0
If set to 1, apply day-  
light savings rules to  
displayed time.  
tcpIpApp.sntp.daylightSavings.fixedDay-  
Enable  
If set to 1, then month  
and date are used (for  
example, April 1st);  
otherwise month, date,  
and dayOfWeek are  
used.  
tcpIpApp.sntp.daylightSavings.start.month  
1-12  
4 (April)  
Month to start DST.  
1=Jan, 2=Feb, ...,  
12=Dec  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
tcpIpApp.sntp.daylightSavings.start.date  
1-31  
1
Day of the month to  
start DST.  
tcpIpApp.sntp.daylightSavings.start.time  
0-23  
2
1
0
Time of day to start  
DST, in 24 hour clock.  
2=2 am, 14=2 pm  
tcpIpApp.sntp.daylightSavings.start.dayOf- 1-7  
Week  
Day of week to apply  
DST. 1=Sun, 2=Mon,  
..., 7=Sat  
tcpIpApp.sntp.daylightSavings.start.dayOf- 0, 10  
Week.lastInMonth  
If set to 1 and fixedDay-  
Enable=0, start DST on  
the last day of the week  
(specified by dayOf-  
Week) in the month,  
rather than the first in  
the month.  
tcpIpApp.sntp.daylightSavings.stop.month  
1-12  
10  
Month to stop DST.  
1=Jan, 2=Feb, ...,  
12=Dec  
tcpIpApp.sntp.daylightSavings.stop.date  
tcpIpApp.sntp.daylightSavings.stop.time  
1-31  
0-23  
1
2
Day of the month to  
start DST.  
Time of day to stop  
DST, in 24 hour clock.  
2= 2 am, 14=2 pm  
tcpIpApp.sntp.daylightSavings.stop.dayOf- 1-7  
Week  
1
1
Day of week to stop  
DST. 1=Sun, 2=Mon,  
..., 7=Sat  
tcpIpApp.sntp.daylightSavings.stop.dayOf- 0, 1  
Week.lastInMonth  
If set to 1 and fixedDay-  
Enable=0, stop DST on  
the last day of the week  
(specified by dayOf-  
Week) in the month,  
rather than the first in  
the month.  
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4.6.1.10.3 port <port/>  
4.6.1.10.3.1 RTP <RTP/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
tcpIpApp.port.rtp.filterByIp  
0, 1  
1
If set to 1, reject RTP pack-  
ets arriving from (sent from)  
a non-negotiated (through  
SDP) IP address.  
tcpIpApp.port.rtp.filterByPort  
tcpIpApp.port.rtp.forceSend  
0, 1  
0
If set to 1, reject RTP pack-  
ets arriving from (sent from)  
a non-negotiated (through  
SDP) port.  
Null, 1024-  
65534  
Null  
When non-Null, send all  
RTP packets to, and expect  
all RTP packets to arrive on,  
the specified port.  
Note: both tcpI-  
pApp.port.rtp.filterByIp and  
tcpIpApp.port.rtp.filterBy-  
Port must be enabled for this  
to work.  
tcpIpApp.port.rtp.mediaPortRangeStart  
Null, even  
integer from  
1024-65534  
Null  
If set to Null, the value 2222  
will be used for the first allo-  
cated RTP port, otherwise,  
the specified port will be  
used. Subsequent ports will  
be allocated from a pool  
starting with the specified  
port plus two up to a value  
of (start-port + 46), after  
which the port number will  
wrap back to the starting  
value.  
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4.6.1.11 Web Server <HTTPD/>  
The phone contains a local web server for user and administrator features. This can be  
disabled for applications where it is not needed or where it poses a security threat. The  
web server supports both basic and digest authentication. The authentication user  
name and password are not configurable for this release.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
httpd.enabled  
0, 1  
1
If set to 1, the HTTP server will be enabled.  
4.6.1.11.1 Configuration <cfg/>  
Permitted  
Values  
Default  
Interpretation  
httpd.cfg.enabled  
0, 1  
1
If set to 1, the HTTP server configuration  
interface will be enabled.  
httpd.cfg.port  
1-65535  
80  
Port is 80 for HTTP servers. Care should be  
taken when choosing an alternate port.  
4.6.1.12 Call Handling Configuration <call/>  
Permitted  
Values  
Attribute  
Default Interpretation  
call.rejectBusyOnDnd  
0, 1  
1
If set to 1, reject all incoming calls with  
the reason “busy” if do-not-disturb is  
enabled.  
call.enableOnNotRegistered  
0, 1  
1
If set to 1, calls will be allowed when the  
phone is not successfully registered, other-  
wise, calls will not be permitted without a  
valid registration.  
call.offeringTimeOut  
call.ringBackTimeOut  
positive  
integer  
60  
60  
Time in seconds to allow an incoming call  
to ring before dropping the call, 0=infi-  
a
nite .  
positive  
integer  
Time in seconds to allow an outgoing call  
to remain in the ringback state before  
dropping the call, 0=infinite.  
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Permitted  
Values  
Attribute  
Default Interpretation  
call.lastCallReturnString  
string of  
maximum  
length 32  
*69  
The string sent to the server when the user  
selects the “last call return” action.  
®
call.callsPerLineKey  
1 to 24 OR 24 OR  
1 to 8 8  
For the SoundPoint IP 600 and 601 the  
permitted range is 1 to 24 and the default  
is 24. For all other phones the permitted  
range is 1 to 8 and the default is 8.  
This is the number of calls that may be  
active or on hold per line key on the  
phone.  
Note that this may be overridden by the  
per-registration attribute of reg.x.callsPer-  
LineKey. Refer to 4.6.2.1 Registration  
<reg/> on page 149.  
call.stickyAutoLineSeize  
0 or 1  
0
Set to 1 to make the phone use "sticky"  
line seize behavior. This will help with  
features that need a second call object to  
work with. The phone will attempt to ini-  
tiate a new outgoing call on the same SIP  
line that is currently in focus on the LCD  
(this was the behavior in SIP 1.6.5).  
Set to 0 means disabled (this was the  
behavior in SIP 1.6.6).  
Note: This may fail due to glare issues in  
which case the phone may select a differ-  
ent available line for the call.  
a. The call diversion, no answer feature will take precedence over this feature if enabled. For  
more information, refer to 4.6.2.3.3 No Answer <noanswer/> on page 155.  
4.6.1.12.1 Shared Calls <shared/>  
Permitted  
Values  
Attribute  
Default Interpretation  
a
0, 1  
1
If set to 1, disable diversion feature for  
shared lines.  
call.shared.disableDivert  
call.shared.seizeFailReorder  
0, 1  
1
If set to 1, play re-order tone locally on  
shared line seize failure.  
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Permitted  
Values  
Attribute  
Default Interpretation  
call.shared.oneTouchResume  
0, 1  
0
Note: This parameter affects the  
®
SoundStation IP 4000 phone only. For  
other phones a quick press and release of  
the line key will resume a call whereas  
pressing and holding down the line key  
will show a list of calls on that line.  
If set to 1, when a shared line has a call  
on hold the remote user can press that  
line and resume the call. If more than one  
call is on hold on the line then the first  
one will be selected and resumed auto-  
matically.  
If set to 0, pressing the shared line will  
bring up a list of the calls on that line and  
the user can select which call the next  
action should be applied to.  
call.shared.exposeAutoHolds  
0, 1  
0
If set to 1, on a shared line, when setting  
up a conference, a re-INVITE will be sent  
to the server.  
If set to 0, no re-INVITE will be sent to  
the server.  
a. This feature is disabled on most call servers.  
4.6.1.12.2 Hold, Local Reminder <hold/><localReminder/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
call.hold.localReminder.enabled  
0, 1  
0
If set to 1, periodically notify the  
local user that calls have been on  
hold for an extended period of  
time.  
call.hold.localReminder.period  
non-negative 60  
integer  
Time in seconds between subse-  
quent reminders.  
call.hold.localReminder.startDelay  
non-negative 90  
integer  
Time in seconds to wait before  
the initial reminder.  
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4.6.1.13 Directory <directory/>  
The directory is stored in either flash memory or RAM on the phone. The directory  
8
size is limited based on the amount of flash memory in the phone .  
When the volatile storage option is enabled, ensure that a properly configured boot  
server that allows uploads is available to store a back-up copy of the directory or its  
contents will be lost when the phone reboots or loses power.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
dir.local.volatile.2meg  
0, 1  
0
Attribute applies to platforms  
with 2 Mbytes of flash mem-  
ory.  
If set to 1, use volatile storage  
for phone-resident copy of the  
directory to allow for larger  
size.  
dir.local.nonVolatile.maxSize.2meg  
1 to 20  
20  
Attribute applies to platforms  
with 2 Mbytes of flash mem-  
ory. Maximum size in Kbytes  
of non-volatile storage that the  
directory will be permitted to  
consume.  
dir.local.volatile.4meg  
0, 1  
0
Applies to platforms with 4  
Mbytes of flash memory.  
If set to 1, use volatile storage  
for phone-resident copy of the  
directory to allow for larger  
size.  
dir.local.nonVolatile.maxSize.4meg  
dir.local.volatile.maxSize  
1 to 50  
50  
Applies to platforms with 4  
Mbytes of flash memory. Max-  
imum size in Kbytes of non-  
volatile storage that the direc-  
tory will be permitted to con-  
sume.  
1 to 100  
100  
Maximum size in Kbytes of  
volatile storage that the direc-  
tory will be permitted to con-  
sume.  
8. Different phone models have variable flash memory.  
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4.6.1.14 Presence <presence/>  
The parameter pres.reg is the line number used to send SUBSCRIBE. If this parameter  
is missing, the phone will use the primary line to send SUBSCRIBE.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
pres.reg  
positive  
integer  
1
Specifies the line/registration  
number used to send SUB-  
SCRIBE for presence. Must be  
a valid line/registration num-  
ber. If the number is not a valid  
line/registration number, it is  
ignored.  
4.6.1.15 Fonts <font/>  
®
This section does not apply to the SoundPoint IP 300 and 301 phones.  
These settings control the phone’s ability to dynamically load an external font file dur-  
ing boot up. Loaded fonts can either overwrite pre-existing fonts embedded within the  
software (not recommended) or can extend the phone’s font support for Unicode  
9
ranges not already embedded. The font file must be a Microsoft .fnt or .fon file for-  
mat. The font file name must follow a specific pattern as described:  
• Font file name: <fontName>_<fontHeightInPixels>_<fontRange>.<fontExtension>  
• <fontName> is a free string of characters that typically carries the meaning of  
the font. Examples are “fontFixedSize” for a fixed-size font, or “fontPropor-  
tionalSize” for a proportional size font.  
• <fontHeightInPixels> describes the font height in number of screen pixels.  
• <fontRange> describes the Unicode range covered by this font. Since .fnt or  
.fon are 256 characters based blocks, the <fontRange> is Uxx00_UxxFF (.fnt  
file) or Uxx00_UyyFF (.fon file). For more information, refer to 3.5.1 Multilin-  
gual User Interface on page 61.  
• <fontExtension> describes the file type. Either .fnt for single 256 characters  
font or .fon for multiple .fnt files.  
9. .fon file format is a collection of .fnt fonts grouped together within a single file.  
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If it is necessary to overwrite an existing font, use these <fontName>_<fontHeightIn-  
Pixels>:  
SoundPoint® IP 430, 500 and  
501  
“fontProp_10”  
This is the font used widely in the current implementation.  
This is the soft key specific font.  
“fontPropSoftkey_10”  
SoundPoint® IP 600 and 601  
“fontProp_19”  
This is the font used widely in the current implementation including  
for soft keys.  
“fontProp_26”  
“fontProp_x”  
This is the font used to display time (but not date).  
This is a small font used for the CPU/Load/Net utilization graphs,  
this is the same as the “fontProp_10” for the SoundPoint® IP 500.  
If the <fontName>_<fontHeightInPixels> does not match any of the names above,  
then the downloaded font will be applied against all fonts defined in the phone, which  
means that you may lose the benefit of fonts being calibrated differently depending on  
their usage. For example, the font used to display the time on the Sound Point® IP 600  
is a large font, larger than the one used to display the date, and if you overwrite this  
default font with a unique font, you lose this size aspect.  
Example of use:  
• to overwrite the font used for SoundPoint® IP 500 soft keys for ASCII, the  
name should be “fontPropSoftkey_10_U0000_U00FF.fnt”  
• to add support for a new font that will be used everywhere and that is not cur-  
rently supported. For example, for the Eastern/Central European Czech lan-  
guage, this is Unicode range 100-17F, the name could be  
“fontCzechIP500_10_U0100_U01FF.fnt” and  
“fontCzechIP600_19_U0100_U01FF.fnt”  
When defining a single .fon file, there is a need for a “font delimiter”, currently  
“Copyright Polycom Canada Ltd” is used as an embedded delimiter, but this can be  
configured using “font.delimiter”. The font delimiter is important to retrieve the dif-  
ferent mangled .fnt blocks. This font delimiter must be placed in the “copyright”  
attribute of the .fnt header. .fon files are useful if you want to include support for a  
large number of font ranges at once, otherwise, if simply adding or changing a few  
fonts currently in use, multiple .fnt files are recommended since they are easier to  
work with individually.  
Attribute  
Permitted Values  
Default  
Interpretation  
font.delimiter  
string up to 256 ASCII  
characters  
Null  
Delimiter required to retrieve differ-  
ent grouped .fnt blocks.  
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4.6.1.15.1 IP_400 font <IP_400/>  
Attribute  
Permitted Values  
Default  
Interpretation  
font.IP_400.x.name fontName_height_Uxx00 Null  
_UyyFF.fon OR  
Defines the font file that will be  
loaded from boot server during boot  
up.  
fontName_height_Uxx00  
_UxxFF.fnt  
Note: When several  
font.IP_430.x.name are defined, the  
index x must follow consecutive  
increasing order.  
4.6.1.15.2 IP_500 font <IP_500/>  
Attribute  
Permitted Values  
Default  
Interpretation  
font.IP_500.x.name fontName_height_Uxx00 Null  
_UyyFF.fon OR  
Defines the font file that will be  
loaded from boot server during boot  
up.  
fontName_height_Uxx00  
_UxxFF.fnt  
Note: When several  
font.IP_500.x.name are defined, the  
index x must follow consecutive  
increasing order.  
4.6.1.15.3 IP_600 font <IP_600/>  
Attribute  
Permitted Values  
Default  
Interpretation  
font.IP_600.x.name  
fontName_height_Uxx  
00_UyyFF.fon OR  
fontName_height_Uxx  
00_UxxFF.fnt  
Null  
Defines the font file that will be  
loaded from boot server during boot  
up.  
Note: When several  
font.IP_600.x.name are defined, the  
index x must follow consecutive  
increasing order.  
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4.6.1.16 Keys <keys/>  
These settings control the scrolling behavior of keys and can be used to change key  
functions.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
key.scrolling.timeout  
positive  
integer  
1
The time-out after which a key that is enabled  
for scrolling will go into scrolling mode until the  
key is released. Keys enabled for scrolling are  
menu navigation keys (left, right, up, down  
arrows), volume keys, and some context-spe-  
cific soft keys. The value is an integer multiple  
of 500 milliseconds (1=500ms).  
SoundPoint® IP 300, 301, 430, 500, 501 and 600 key functions can be changed from  
the factory defaults, although this is typically not necessary. For each key whose func-  
tion you wish to change, add an XML attribute in the format described in the following  
table to the <keys .../> element of the configuration file. These will override the built-  
in assignments.  
Remapping the arrow keys is not recommended.  
In the following table, x=IP_300, IP 430, IP_500 or IP_600, y is the key number. Note  
that IP_300 parameters affect SoundPoint® IP 300 and 301 phones, IP_430 parameters  
affect SoundPoint® IP 430 phones, and IP_500 parameters affect SoundPoint® IP 500  
and 501 phones. IP 300: y=1-35; IP 430: y=1-35; IP 500: y=1-40; IP 600: y=1-42  
Permitted  
Values  
Attribute  
Interpretation  
key.x.y.function.prim  
Functions listed  
below.  
Sets the function for key y on platform x.  
key.x.y.subPoint.prim positive integer  
Sets the sub-identifier for key functions with a  
secondary array identifier such as SpeedDial.  
The following table lists the functions that are available:  
Function  
Function  
Line1  
Line2  
Line3  
Line4  
ArrowDown  
ArrowLeft  
ArrowRight  
ArrowUp  
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Function  
Function  
Line5  
BuddyStatus  
CallList  
Line6  
Conference  
Delete  
Messages  
Menu  
Dialpad0  
Dialpad1  
Dialpad2  
Dialpad3  
Dialpad4  
Dialpad5  
Dialpad6  
Dialpad7  
Dialpad8  
Dialpad9  
DialpadStar  
DialpadPound  
Directories  
DoNotDisturb  
Handsfree  
Headset  
MicMute  
MyStatus  
Null  
Offline  
Redial  
Select  
Setup  
SoftKey1  
SoftKey2  
SoftKey3  
SoftKey4  
SpeedDial  
SpeedDialMenu  
Transfer  
VolDown  
VolUp  
Hold  
4.6.1.17 Bitmaps <bitmaps/>  
Bitmaps used by the phone are defined in this section.  
4.6.1.17.1 Platform <IP_300/>, <IP_400/>, <IP_500/>, <IP_600/>  
and <IP_4000/>  
In the following table, x=IP_300, IP_400, IP_500, IP_600, or IP_4000 and y is the bit-  
map number. Note that IP_300 parameters affect SoundPoint® IP 300 and 301 phones,  
IP_400 parameters affects SoundPoint® IP 430 phones, IP_500 parameters affect  
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SoundPoint® IP 500 and 501 phones and IP_600 parameters affect SoundPoint® IP 600  
and 601 phones.  
Attribute  
Permitted Values  
Interpretation  
bitmap.x.y.name  
The name of a bit-  
map to be used.  
This is the name of a bitmap to be used for creating an  
animation. If the bitmap is to be downloaded from the  
boot server, its name must:  
1. Be different from any name already in use in  
sip.cfg.  
2. Match the name of the corresponding <file-  
Name>.bmp to be retrieved from the boot server.  
4.6.1.18 Indicators <indicators/>  
Indicators (graphic icons, animations, and LED patterns) used by the phone are  
defined in this section.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
ind.idleDisplay.enabled  
0, 1  
0
If set to 1, the idle display may support pre-  
sentation of a custom animation if config-  
ured properly in the animation section of  
sip.cfg.  
4.6.1.18.1 Animations <Animations/> <IP_300/>, <IP_400/>,  
<IP_500/>, <IP_600/> and <IP_4000/>  
This section defines bitmap animations composed of bitmap/duration couples. In the  
following table, x=IP_300, IP_400, IP_500, IP_600 or IP_4000, y is the animation  
number, z is the step in the animation. Note that IP_300 parameters affect SoundPoint®  
IP 300 and 301 phones, IP_400 parameters affect SoundPoint® IP 430 phones, IP_500  
parameters affect SoundPoint® IP 500 and 501 phones and IP_600 parameters affect  
SoundPoint® IP 600 and 601 phones.  
Attribute  
Permitted Values  
Interpretation  
ind.anim.x.y.frame.z.bitmap  
A bitmap name  
Bitmap to use.  
defined previously.  
Note that it must be defined already, refer  
to 4.6.1.17.1 Platform <IP_300/>,  
<IP_400/>, <IP_500/>, <IP_600/> and  
<IP_4000/> on page 133.  
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Attribute  
Permitted Values  
Interpretation  
ind.anim.x.y.frame.z.duration  
positive integer  
Duration in milliseconds for this step.  
0=infinite.  
4.6.1.18.2 Patterns <Patterns/>  
This section defines patterns for the LED indicators. In the following table, x is the  
pattern number, y is the step in the pattern.  
Permitted  
Values  
Attribute  
Interpretation  
ind.pattern.x.step.y.state  
ind.pattern.x.step.y.duration  
ind.pattern.x.step.y.colour  
On or Off  
Turn LED on or off for this step.  
positive integer Duration in milliseconds for this step. 0=infinite  
Red or Green  
(default is Red  
if not specified)  
For bi-color LEDs, specify color.  
4.6.1.18.3 Classes <Classes/>  
This section defines the available classes for the LED and graphical icon indicator  
types. In the following table, x is the class number, y is the identifier of the state num-  
ber for that class.  
Permitted  
Values  
Attribute  
Interpretation  
ind.class.x.state.y.index  
positive integer For LED type indicators, index refers to the pattern  
index, such as index x in the <Patterns/> tag above.  
For GraphicIcon type indicators, index refers to the  
animation index, such as index y in the <Anima-  
tions/> tag above.  
4.6.1.18.4 Assignments <Assignments/>  
This section assigns a type, a class, and, in the case of the GraphicIcon type, a physical  
location and size in pixels on the LCD display or in the case of the LED type, a physi-  
cal LED number.  
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4.6.1.18.4.1 LEDs <led/>  
In the following table, x is the LED number.  
Permitted  
Values  
Attribute  
Interpretation  
ind.led.x.index  
This is for internal usage only and should not be changed  
(this is the logical index).  
ind.led.x.class  
positive integer Assigns the class (defined in 4.6.1.18.3 Classes <Classes/  
> on page 135) for this indicator.  
ind.led.x.physNum  
This maps the logical index to a specific physical LED.  
4.6.1.18.4.2 Graphic Icons <gi/> <IP_300/>, <IP_400/>, <IP_500/>, <IP_600/>  
and <IP_4000/>  
In the following table, x=IP_300, IP_400, IP_500, IP_600 or IP_4000, y is the graphic  
icon number. Note that IP_300 parameters affect SoundPoint® IP 300 and 301 phones,  
IP_400 parameters affect SoundPoint® IP 430 phones, IP_500 parameters affect  
SoundPoint® IP 500 and 501 phones, and IP_600 parameters affect SoundPoint® IP  
600 and 601 phones.  
Permitted  
Values  
Attribute  
Interpretation  
ind.gi.x.y.index  
This is for internal usage only and should not be changed  
(this is the logical index).  
ind.gi.x.y.class  
positive integer Assigns the class (defined in 4.6.1.18.3 Classes <Classes/  
> on page 135) for this indicator.  
ind.gi.x.y.physX  
IP 300: 0-19  
For GraphicIcon type indicators, this is the x-axis loca-  
tion of the upper left corner of the indictor measured in  
pixels from left to right.  
IP 400: 0-122  
IP 500: 0-159  
IP 600: 0-319  
IP 4000: 0-247  
ind.gi.x.y.physY  
IP 300: 0-3  
For GraphicIcon type indicators, this is the y-axis loca-  
tion of the upper left corner of the indicator measured in  
pixels from top to bottom.  
IP 400: 0-45  
IP 500: 0-79  
IP 600: 0-159  
IP 4000: 0-67  
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Permitted  
Values  
Attribute  
Interpretation  
ind.gi.x.y.physW  
IP 300: n/a  
For GraphicIcon type indicators, this is the width of the  
indicator measured in pixels.  
IP 400: 1-94  
IP 500: 1-160  
IP 600: 1-320  
IP 4000: 1-248  
ind.gi.x.y.physH  
IP 300: n/a  
For GraphicIcon type indicators, this is the height of the  
indicator measured in pixels.  
IP 400: 1-23  
IP 500: 1-80  
IP 600: 1-160  
IP 4000: 1-68  
4.6.1.19 Event Logging <logging/>  
Important  
Logging parameter changes can impair system operation. Do not change any logging parameters with-  
out prior consultation with Polycom Customer Support.  
The event logging system supports the following classes of events:  
Level  
Interpretation  
0
1
2
3
4
5
6
Debug only  
High detail event class  
Moderate detail event class  
Low detail event class  
Minor error - graceful recovery  
Major error - will eventually incapacitate the system  
Fatal error  
Each event in the log contains the following fields separated by the | character:  
• time or time/date stamp  
• 1-5 character component identifier (such as “so”)  
• event class  
• cumulative log events missed due to excessive CPU load  
• free form text - the event description  
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Example:  
011511.006|so  
|2|00|soCoreAudioTermChg: chassis -> idle  
time stamp  
ID  
event class  
missed events  
text  
Three formats are available for the event timestamp:  
Type  
Example  
0 - seconds.milliseconds  
011511.006-- 1 hour, 15 minutes, 11.006 seconds  
since booting.  
1 - absolute time with minute resolution  
2 - absolute time with seconds resolution  
0210281716-- 2002 October 28, 17:16  
1028171642-- October 28, 17:16:42  
4.6.1.19.1 Basic Logging <level/><change/> and <render/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
log.level.change.xxx  
0-5  
4
Control the logging detail level  
for individual components.  
These are the input filters into  
the internal memory-based log  
system.  
log.render.level  
0-6  
1
Specifies the lowest class of  
event that will be rendered to  
the log files. This is the output  
filter from the internal mem-  
ory-based log system.  
log.render.type  
0-2  
2
1
Refer to above table for times-  
tamp type.  
log.render.realtime  
0, 1  
Set to 1.  
Note: Polycom recommends  
that you do not change this  
value.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
log.render.stdout  
0, 1  
1
Set to 1.  
Note: Polycom recommends  
that you do not change this  
value.  
log.render.file  
0, 1  
1
Set to 1.  
Note: Polycom recommends  
that you do not change this  
value.  
log.render.file.size  
positive  
integer, 1 to  
179.5  
16  
Maximum local application  
log file size in Kbytes. When  
this size is exceeded, the file is  
uploaded to the boot server  
and the local copy is erased.  
log.render.file.upload.period  
positive  
integer  
172800  
Time in seconds between log  
file uploads to the boot server.  
Note: The log file will not be  
uploaded if no new events  
have been logged since the last  
upload.  
log.render.file.upload.append  
0, 1  
1
If set to 1, use append mode  
when uploading log files to  
server.  
Note: HTTP and TFTP don’t  
support append mode unless  
the server is set up for this.  
log.render.file.upload.append.sizeLimit  
positive  
integer  
512  
Maximum log file size on boot  
server in Kbytes.  
log.render.file.upload.append.limit-  
Mode  
delete, stop  
delete  
Behavior when server log file  
has reached its limit.  
delete=delete file and start  
over  
stop=stop appending to file  
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4.6.1.19.2 Scheduled Logging Parameters <scheduled/>  
The phone can be configured to schedule certain advanced logging tasks on a periodic  
basis. These attributes should be set in consultation with Polycom. Each scheduled log  
task is controlled by a unique attribute set starting with log.sched.x where x identifies  
the task.  
Permitted  
Values  
Attribute  
Interpretation  
log.sched.x.name  
alphanu-  
Name of an internal system command to be periodically  
meric string executed. To be supplied by Polycom.  
log.sched.x.level  
log.sched.x.period  
0-5  
Event class to assign to the log events generated by this  
command. This needs to be the same or higher than  
log.level.change.slog for these events to appear in the log.  
positive  
integer  
Seconds between each command execution. 0=run once  
log.sched.x.startMode abs, rel  
Start at absolute time or relative to boot.  
log.sched.x.startTime  
positive  
integer OR  
hh:mm  
Seconds since boot when startMode is rel or the start time  
in 24-hour clock format when startMode is abs.  
log.sched.x.startDay  
1-7  
When startMode is abs, specifies the day of the week to  
start command execution. 1=Sun, 2=Mon, ..., 7=Sat  
4.6.1.20 Security <security/>  
These settings affect security aspects of the phone.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
sec.tagSerialNo  
0, 1  
0
If set to 1, the phone may advertise its serial num-  
ber (Ethernet address) through protocol signaling.  
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4.6.1.20.1 Encryption <encryption/>  
Permitted  
Optimization  
Attribute  
Values  
Default  
Interpretation  
sec.encryp-  
tion.upload.dir  
0, 1  
0
If set to 0, the phone-specific contact direc-  
tory is uploaded to the server unencrypted  
regardless of how it was downloaded. This  
will replace whatever phone-specific contact  
directory is on the server even if it is  
encrypted.  
If set to 1, the phone-specific contact direc-  
tory is uploaded encrypted regardless of  
how it was downloaded. This will replace  
whatever phone-specific contact directory is  
on the server even if it is unencrypted.  
sec.encryp-  
tion.upload.overrides  
0, 1  
0
If set to 0, the phone-specific configuration  
override file (<Ethernet Address>-  
phone.cfg) is uploaded unencrypted regard-  
less of how it was downloaded. This will  
replace the override file on the server even if  
it is encrypted.  
If set to 1, the phone-specific configuration  
override file is uploaded encrypted regard-  
less of how it was downloaded. This will  
replace the override file on the server even if  
it is unencrypted.  
4.6.1.20.2 Password Lengths <pwd/><length/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
sec.pwd.length.admin  
sec.pwd.length.user  
0-32  
1
2
Password changes will need to be at least  
this long. Use 0 to allow null passwords.  
0-32  
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4.6.1.21 Provisioning <provisioning/>  
These settings control aspects of the phone’s boot server provisioning system.  
Permitted  
Attribute  
Values  
Default  
5
Interpretation  
prov.fileSystem.rfs0.minFreeSpace  
prov.fileSystem.ffs0.4meg.minFreeSpace  
prov.fileSystem.ffs0.2meg.minFreeSpace  
5-512  
Important: Polycom recom-  
mends that you do not  
change these parameters.  
420  
48  
Minimum free space in  
Kbytes to reserve in the file  
system when downloading  
files from the boot server.  
prov.polling.enabled  
0, 1  
0
If set to 1, automatic periodic  
boot server polling for  
upgrades is enabled.  
prov.polling.mode  
prov.polling.period  
abs, rel  
abs  
Polling mode is absolute or  
relative.  
integer  
greater  
than 3600  
86400  
Polling period in seconds.  
Rounded up to the nearest  
number of days in abs mode.  
Measured relative to boot  
time in rel mode.  
prov.polling.time  
Format is  
hh:mm  
03:00  
Only used in abs mode. Poll-  
ing time.  
4.6.1.22 RAM Disk <RAMdisk/>  
These settings control the phone’s internal RAM disk feature. Changing these parame-  
ters is not advised.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
ramdisk.enable  
0, 1  
1
If set to 1, RAM disk will be available. The  
RAM disk is used to cache downloaded  
wave files, and other resources for the user  
interface.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
ramdisk.bytesPerBlock  
0, 32, 33, ...,  
1024  
0
These three parameters use internal defaults  
when value is set to 0.  
ramdisk.blocksPerTrack  
ramdisk.nBlocks  
0, 1, 2, ...,  
65536  
0
0, 1, 2, ...,  
65536  
4096  
50  
ramdisk.minsize  
50 to 16384  
Smallest size in Kbytes of RAM disk to cre-  
ate before returning an error. RAM disk size  
is variable depending on the amount of  
device memory.  
ramdisk.minfree  
512 to  
16384  
3072  
Minimum amount of free space that must be  
left after the RAM disk has been created.  
The RAM disk’s size will be reduced as  
necessary in order to leave this amount of  
free RAM.  
4.6.1.23 Request <request/>  
4.6.1.23.1 Delay <delay/>  
These settings control the phone’s behavior when a request for restart, reboot, or  
reconfiguration is received.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
request.delay.type Null,  
“audio”, or  
“call”  
call  
Defines the strategy to adopt before a request gets  
executed. If set to “audio”, a request can be exe-  
cuted as soon as there is no active audio on the  
phone, independently of any call state. If set to  
“call”, a request can be executed as soon as there  
are no calls in any state on the phone.  
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4.6.1.24 Feature <feature/>  
These settings control the activation or deactivation of a feature at run time. In the  
table below, x is the feature number.  
Attribute  
Permitted Values  
Interpretation  
feature.x.name  
“presence”,  
These are features offered on the phone:  
“messaging”,  
“directory”,  
“calllist”,  
“ring-download”,  
“calllist-received”,  
“calllist-placed”,  
“calllist-missed”,  
“url-dialing”,  
•
“presence” is the presence feature including  
management of buddies and own status  
“messaging” is the instant messaging feature  
“directory” is the local directory feature  
“calllist” is the locally controlled call lists  
“ring-download” is run-time downloading of  
ringers  
“calllist-received” is the received-calls list fea-  
ture (the “calllist” feature must be enabled for  
this feature to be available)  
“calllist-placed” is the placed-calls list feature  
(the “calllist” feature must be enabled for this  
feature to be available)  
“calllist-missed” is the missed-calls list feature  
(the “calllist” feature must be enabled for this  
feature to be available)  
•
•
•
•
•
•
•
•
“call-park”,  
“group-call-pickup”,  
“directed-call-pickup”,  
“last-call-return”,  
“acd-login-logout”,  
“acd-agent-available”  
“url-dialing” controls whether URL/name dial-  
ing is available from a private line (it is never  
available from a shared line)  
•
•
•
“call-park” is the call park and park-retrieve  
features  
“group-call-pickup” is the group call pickup  
feature  
“directed-call-pickup” is the directed call  
pickup feature  
•
•
“last-call-return” is the last call return feature  
“acd-login-logout” is the ACD login/logout fea-  
ture  
•
“acd-agent-available” is the ACD agent avail-  
able/unavailable feature  
feature.x.enabled 0 or 1 (default) except  
for x=9  
If set to 0, the feature will be disabled.  
If set to 1, the feature will be enabled and usable by  
the local user.  
Note: The "url-dialing" feature must be disabled by  
setting feature.9.enabled to 0 in order to prevent  
unknown callers from being identified on the display  
by an IP address.  
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4.6.1.25 Resource <resource/>  
These settings control the maximum size or an external resource retrieved at run time.  
4.6.1.25.1 finder <finder/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
res.finder.sizeLimit  
positive  
integer  
300  
If a resource that is being downloaded to the  
phone is larger than this value * 1000 bytes (=  
the maximum size), the resource will be auto-  
matically truncated to the maximum size  
defined.  
res.finder.minfree  
1 to 2048  
1200  
Used to ensure that the phone will not down-  
load resources which could leave it with  
insufficient memory to function correctly. A  
resource is not be downloaded if the phone  
has less memory free than res.finder.minFree  
in kBytes. The recommended value is 1200. If  
the parameter is left empty, the default is 800.  
Note: Setting this value too small may affect  
functionality of the phone. Setting this value  
too large may mean that some resources are  
not downloaded at boot time.  
4.6.1.25.2 quotas <quotas/>  
Permitted  
Values  
Attribute  
Interpretation  
res.quotas.x.name  
“tone”, “bit-  
The name of the sub-application for which the particu-  
map”, or “font” lar quota will apply:  
•
“tone” relates to all downloaded tones and sound  
effects  
•
•
“bitmap” relates to all downloaded bitmaps  
“font” relates to all downloaded fonts  
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Permitted  
Values  
Attribute  
Interpretation  
res.quotas.x.value  
positive integer  
When resources that fall in the defined category are  
downloaded to the phone, a quota equal to this value *  
1024 bytes of compound data size is applied for that  
category. If downloading a resource would make the  
quota exceeded for that category, the resource will not  
be downloaded and a predefined default will be used  
instead.  
For res.quotas.tone.value: default is 600 KB for tones,  
10 KB for bitmaps and fonts.  
4.6.1.26 MicroBrowser <microbrowser/>  
These settings control the home page, proxy and size limits to be used by the Micro-  
Browser when it is selected to provide services.  
Attribute  
Permitted Values  
Default  
Interpretation  
mb.proxy  
Null or  
Null.  
Address of the desired HTTP proxy to be  
used by the MicroBrowser. If blank, nor-  
mal unproxied HTTP is used by the  
MicroBrowser.  
domain name or  
IP address in the  
format  
Default  
port =  
8080  
<address>:<port>  
4.6.1.26.1 Idle Display <idleDisplay/>  
The MicroBrowser can be used to create a display that will be part of the phone’s idle  
display. These settings control the home page and the refresh rate.  
Attribute  
Permitted Values  
Default  
Interpretation  
mb.idleDisplay.home Null or any fully  
formed valid  
Null  
URL used for MicroBrowser idle display  
home page. example: http://www.exam-  
ple.com/xhtml/  
HTTP URL.  
Length up to 255  
characters.  
frontpage.cgi?page=home. If empty,  
there will be no MicroBrowser idle dis-  
play feature. Note that the MicroBrowser  
idle display will displace the idle display  
indicator (refer to ind.idleDisplay.enabled  
in 4.6.1.18 Indicators <indicators/> on  
page 134).  
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Attribute  
Permitted Values  
Default  
Interpretation  
mb.idleDis-  
play.refresh  
0 or an integer > 5  
0
The period in seconds between refreshes  
of the idle display MicroBrowser’s con-  
tent. If set to 0, the idle display Micro-  
Browser is not refreshed. The minimum  
refresh period is 5 seconds (values from 1  
to 4 are ignored, and 5 is used).  
Note: If an HTTP Refresh header is  
detected, it will be respected, even if this  
parameter is set to 0. The use of this  
parameter in combination with the  
Refresh HTTP header may cause the idle  
display to refresh at unexpected times.  
4.6.1.26.2 Main Browser <main/>  
This setting controls the home page used by the MicroBrowser when that function is  
selected.  
Attribute  
Permitted Values  
Default  
Interpretation  
mb.main.home  
Any fully formed  
valid HTTP URL.  
Length up to 255  
characters.  
Null  
URL used for MicroBrowser home-page.  
If blank, the browser will notify the user  
that a blank home-page was used.  
Example: http://www.example.com/  
xhtml/frontpage.cgi?page=home.  
4.6.1.26.3 Browser Limits <limits/>  
These settings limit the size of object which the MicroBrowser will display by limiting  
the amount of memory available for the MicroBrowser.  
Attribute  
Permitted Values  
Default  
Interpretation  
mb.limits.nodes  
positive integer  
256  
Limits the number of tags which the  
XML parser will handle. This limits the  
amount of memory used by complicated  
pages. A maximum total of 500 (256  
each) is recommended. This value is used  
as referent values for 16MB of SDRAM.  
Note: Increasing this value may have a  
detrimental effect on performance of the  
phone.  
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Attribute  
Permitted Values  
Default  
Interpretation  
mb.limits.cache  
positive integer  
200  
Limits the total size of objects down-  
loaded for each page (both XHTML and  
images). Once this limit is reached, no  
more images are downloaded until the  
next page is requested. Units = kBytes.  
This value is used as referent values for  
16MB of SDRAM.  
Note: Increasing this value may have a  
detrimental effect on performance of the  
phone.  
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4.6.2 Per-phone Configuration - phone1.cfg  
This section covers the parameters in the per-phone example configuration file  
phone1.cfg. This file would normally be used as a template for the per-phone configu-  
ration files. For more information, refer to 2.2.2.1.2 Boot Server Deployment for the  
Phones on page 19.  
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.  
4.6.2.1 Registration <reg/>  
SoundPoint® IP 300, 301, and 430 support a maximum of two unique registrations,  
SoundPoint® IP 500 and 501 support three, SoundPoint® IP 600 supports six, and  
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 registra-  
tions. Each registration can optionally be associated with a private array of servers for  
completely segregated signaling. SoundStation® IP 4000 supports a single registration.  
In the following table, x is the registration number. IP 300, 301, and 430: x=1-2; IP  
500 and 501: x=1-3; IP 600: x=1-6; IP 601: x=1-12; IP 4000: x=1.  
Permitted  
Values  
Attribute  
Default Interpretation  
reg.x.displayName  
UTF-8 encoded  
string  
Null  
Display name used for local user inter-  
face as well as SIP signaling.  
reg.x.address  
string in the for- Null  
mat userPart or  
from user-  
The user part or the user and the host  
part of the phone’s SIP URI.  
The user part of the phone's SIP URI.  
For example, reg.x.address=”1002”  
from [email protected] or  
Part@domain  
reg.x.address=”[email protected]”.  
reg.x.label  
UTF-8 encoded  
string  
Null  
Text label to appear on the display  
adjacent to the associated line key. If  
omitted, the label will be derived from  
the user part of reg.x.address.  
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Permitted  
Values  
Attribute  
Default Interpretation  
®
reg.x.lcs  
0, 1  
0
If set to 1, the Microsoft Office Live  
Communications Server 2005 is sup-  
ported for registration x.  
reg.x.type  
private OR  
shared  
private If set to private, use standard call sig-  
naling.  
If set to shared, augment call signaling  
with call state subscriptions and notifi-  
cations and use access control for out-  
going calls.  
reg.x.thirdPartyName  
reg.x.auth.userId  
string in the  
same format as  
reg.x.address  
Null  
Null  
This field must match the reg.x.address  
value of the other registration which  
makes up the bridged line.  
string  
User ID to be used for authentication  
challenges for this registration. If non-  
Null, will override the “Reg User x”  
parameter entered into the Authentica-  
tion submenu off of the Settings menu  
on the phone.  
reg.x.auth.password  
string  
Null  
Password to be used for authentication  
challenges for this registration. If non-  
Null, will override the “Reg Password  
x” parameter entered into the Authenti-  
cation submenu off of the Settings  
menu on the phone.  
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Permitted  
Values  
Attribute  
Default Interpretation  
reg.x.server.y.address  
dotted-decimal  
IP address or  
host name  
Null  
Optional IP address or host name, port,  
transport, registration period, fail-over  
parameters and lineseize subscription  
period of a SIP server that accepts reg-  
istrations. Multiple servers can be listed  
starting with y=1, 2, ... for fault toler-  
ance. If specified, these servers will  
override the servers specified in sip.cfg  
in 4.6.1.1.2 Server <server/> on  
page 85.  
reg.x.server.y.port  
0, Null, 1 to  
65535  
Null  
reg.x.server.y.transport  
DNSnaptr or  
DNSna  
TCPpreferred or ptr  
UDPonly or  
TLS  
Note: If the reg.x.server.y.address  
parameter is non-Null, all of the  
reg.x.server.y.xxx parameters will  
override the parameters specified in  
sip.cfg in 4.6.1.1.2 Server <server/> on  
page 85.  
reg.x.server.y.expires  
reg.x.server.y.register  
positive integer  
0, 1  
Null  
Null  
60  
reg.x.server.y.expires.over- positive integer,  
lap  
minimum 5,  
maximum  
65535  
Note: TLS is not supported on Sound-  
®
Point IP 300 and 500 phones.  
reg.x.server.y.retryTime-  
Out  
Null or non-neg- Null  
ative integer  
reg.x.server.y.retryMax-  
Count  
Null or non-neg- Null  
ative integer  
reg.x.server.y.expires.lineS positive integer  
eize  
Null  
reg.x.acd-login-logout  
0, 1  
0, 1  
0
0
If both parameters are set to 1 for a reg-  
istration, the ACD feature will be  
enabled for that registration.  
reg.x.acd-agent-available  
reg.x.ringType  
reg.x.lineKeys  
1 to 22  
2
The ringer to be used for calls received  
by this registration. Default is the first  
non-silent ringer.  
1 to max  
1
max = the number of line keys on the  
phone.  
®
max = 1 on SoundStation IP 4000,  
max = 2 on IP 300, 301, and 430,  
max = 3 on IP 500 and 501,  
max = 6 on IP 600,  
max = 24 on IP 601 (without any  
Expansion Modules attached, only 6  
line keys are available)  
The number of line keys on the phone  
to be associated with registration ‘x’.  
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Permitted  
Values  
Attribute  
Default Interpretation  
®
reg.x.callsPerLineKey  
1 to 24 OR  
1 to 8  
24 OR  
8
For the SoundPoint IP 600 and 601  
the permitted range is 1 to 24 and the  
default is 24.  
For all other phones the permitted  
range is 1 to 8 and the default is 8.  
This is the number of calls or confer-  
ences which may be active or on hold  
per line key associated with this regis-  
tration.  
Note that this overrides call.callsPer-  
LineKey for this registration. Refer to  
4.6.1.12 Call Handling Configuration  
<call/> on page 125.  
reg.x.outbound-  
Proxy.address  
dotted-decimal  
IP address or  
host name  
Null  
IP address or host name and port of a  
SIP server to which the phone shall  
send all requests.  
reg.x.outboundProxy.port  
1 to 65535  
5060  
reg.x.outboundProxy.trans- DNSnaptr or  
DNSna If set to Null or DNSnaptr:  
If reg.x.outboundProxy.address is a  
hostname and reg.x.outbound-  
port  
TCPpreferred or ptr  
UDPonly or  
TLS  
Proxy.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 reg.x.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.  
Note: TLS is not supported on Sound-  
®
Point IP 300 and 500 phones.  
reg.x.proxyRequire  
string  
Null  
The string that needs to appear in the  
“Proxy-Require” header. If Null, no  
"Proxy-Require" will be sent.  
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4.6.2.2 Calls <call/>  
These sections describe call-oriented per-phone configuration items.  
4.6.2.2.1 Do Not Disturb <donotdisturb/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
call.donotdisturb.perReg  
0, 1  
0
If set to 1, the DND feature will allow  
selection of DND on a per-registration  
basis.  
4.6.2.2.2 Automatic Off-hook Call Placement <autoOffHook/>  
An optional per-registration feature is supported which allows automatic call place-  
ment when the phone goes off-hook.  
In the following table, x is the registration number. IP 300, 301, and 430: x=1-2; IP  
500 and 501: x=1-3; IP 600: x=1-6; IP 601: x=1-12; IP 4000: x=1  
Attribute  
Permitted Values  
Default  
Interpretation  
call.autoOffHook.x.enabled  
call.autoOffHook.x.contact  
0, 1  
0
If set to 1, a call  
will be automati-  
cally placed to the  
contact specified  
upon going off  
hook on this regis-  
tration.  
ASCII encoded string containing Null  
digits (the user part of a SIP  
URL) or a string that constitutes  
a valid SIP URL (6416 or  
[email protected])  
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4.6.2.2.3 Missed Call Configuration <serverMissedCall/>  
The phone supports a per-registration configuration of which events will cause the  
locally displayed “missed calls” counter to be incremented.  
In the following table, x is the registration number. IP 300, 301, and 430: x=1-2; IP  
500 and 501: x=1-3; IP 600: x=1-6; IP 601: x=1-12; IP 4000: x=1  
Permitted  
Values  
Attribute  
Default Interpretation  
call.serverMissedCall.x.enabled 0, 1  
0
If set to 0, all missed-call events will  
increment the counter.  
If set to 1, only missed-call events sent  
by the server will increment the counter.  
4.6.2.3 Diversion <divert/>  
The phone has a flexible call forward/diversion feature for each registration. In all  
cases, a call will only be diverted if a non-Null contact has been configured.  
In the following tables, x is the registration number. IP 300, 301, and 430: x=1-2; IP  
500 and 501: x=1-3; IP 600: x=1-6; IP 601: x=1-12; IP 4000: x=1  
Attribute  
Permitted Values  
Default  
Interpretation  
divert.x.contact  
ASCII encoded string  
containing digits (the  
user part of a SIP URL)  
or a string that consti-  
tutes a valid SIP URL  
(6416 or 6416@poly-  
com.com  
Null  
The forward-to contact  
used for all automatic call  
diversion features unless  
overridden by a specific  
contact of a per-call diver-  
sion feature (refer to  
below).  
divert.x.autoOnSpecificCaller 0, 1  
1
1
If set to 1, calls may be  
diverted using the Auto  
Divert feature of the direc-  
tory. This is a global flag.  
divert.x.sharedDisabled  
0, 1  
If set to 1, all diversion fea-  
tures on that line will be  
disabled if the line is con-  
figured as shared.  
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4.6.2.3.1 Forward All <fwd/>  
Permitted  
Optimization  
Attribute  
Values  
Default  
Interpretation  
divert.fwd.x.enabled  
0, 1  
1
If set to 1, the user will be able to enable uni-  
versal call forwarding through the soft key  
menu.  
4.6.2.3.2 Busy <busy/>  
Calls can be automatically diverted when the phone is busy.  
Attribute  
Permitted Values  
Default Interpretation  
divert.busy.x.enabled 0, 1  
1
If set to 1, calls will be  
forwarded on busy to  
the contact specified  
below.  
divert.busy.x.timeout positive integer  
60  
Time in seconds to  
allow altering before  
initiating the diversion.  
divert.busy.x.contact  
ASCII encoded string containing  
Null  
Forward-to contact for  
calls forwarded due to  
busy status, if Null,  
divert.x.contact will be  
used.  
digits (the user part of a SIP URL) or  
a string that constitutes a valid SIP  
URL (6416 or [email protected]  
4.6.2.3.3 No Answer <noanswer/>  
The phone can automatically divert calls after a period of ringing.  
Attribute  
Permitted Values  
Default  
Interpretation  
divert.noanswer.x.enabled  
0, 1  
1
If set to 1, calls will be for-  
warded on no answer to the  
contact specified.  
divert.noanswer.x.timeout  
positive integer  
60  
Time in seconds to allow  
altering before initiating the  
diversion.  
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Attribute  
Permitted Values  
Default  
Interpretation  
divert.noanswer.x.contact  
ASCII encoded string con- Null  
taining digits (the user part  
of a SIP URL) or a string  
that constitutes a valid SIP  
URL (6416 or 6416@poly-  
com.com)  
Forward-to contact used for  
calls forwarded due to no  
answer, if Null,  
divert.x.contact will be  
used.  
4.6.2.3.4 Do Not Disturb <dnd/>  
The phone can automatically divert calls when Do Not Disturb (DND) is enabled.  
Attribute  
Permitted Values  
Default  
Interpretation  
divert.dnd.x.enabled  
0, 1  
0
If set to 1, calls will be for-  
warded on DND to the  
contact specified below.  
divert.dnd.x.contact  
ASCII encoded string containing  
digits (the user part of a SIP URL)  
or a string that constitutes a valid  
SIP URL (6416 or 6416@poly-  
com.com)  
Null  
Forward-to contact used  
for calls forwarded due to  
DND status, if Null  
divert.x.contact will be  
used.  
4.6.2.4 Dial Plan <dialplan/>  
Per-registration dial plan configuration is supported. In the following tables, x is the  
registration number. IP 300, 301, and 430: x=1-2; IP 500 and 501: x=1-3; IP 600: x=1-  
6; IP 601: x=1-12; IP 4000: x=1  
Permitted  
Values  
Attribute  
Default  
Interpretation  
dialplan.x.impossibleMatchHandling 0, 1 or 2  
0
When present, and if dial-  
plan.x.digitmap is not Null,  
this attribute overrides the  
global dial plan defined in  
the sip.cfg configuration file.  
For interpretation, refer to  
4.6.1.2 Dial Plan <dialplan/>  
on page 94.  
156  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
dialplan.x.removeEndOfDial  
0, 1  
1
When present, and if dial-  
plan.x.digitmap is not Null,  
this attribute overrides the  
global dial plan defined in  
the sip.cfg configuration file.  
For interpretation, refer to  
4.6.1.2 Dial Plan <dialplan/>  
on page 94.  
4.6.2.4.1 Digit Map <digitmap/>  
Permitted  
Values  
Attribute  
Default  
Interpretation  
dialplan.x.digitmap  
string compatible with  
the digit map feature of  
MGCP described in  
Null  
When present, this attribute  
overrides the global dial  
plan defined in the sip.cfg  
configuration file.  
2.1.5 of RFC 3435;  
string is limited to 512  
bytes and 20 segments; a  
comma is also allowed;  
when reached in the  
For more information, refer  
to 4.6.1.2.1 Digit Map  
<digitmap/> on page 94.  
digit map, a comma will  
turn dial tone back on.  
dialplan.x.digitmap.timeOut  
positive integer  
Null  
When present, and if dial-  
plan.x.digitmap is not Null,  
this attribute overrides the  
global dial plan defined in  
the sip.cfg configuration  
file.  
For more information, refer  
to 4.6.1.2.1 Digit Map  
<digitmap/> on page 94.  
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4.6.2.4.2 Routing <routing/>  
This configuration section allows specific routing paths for outgoing SIP calls to be  
configured independent of other ‘default’ configuration.  
4.6.2.4.2.1 Server <server/>  
Attribute  
Permitted Values  
Default  
Interpretation  
dialplan.x.rout-  
ing.server.y.address  
dotted-decimal IP  
address or host name  
Null  
IP address or host name  
and port of a SIP server  
that will be used for  
routing calls. Multiple  
servers can be listed  
starting with y=1, 2, ...  
for fault tolerance.  
dialplan.x.rout-  
ing.server.y.port  
1 to 65535  
5060  
4.6.2.4.2.2 Emergency <emergency/>  
In the following attributes, y is the index of the emergency entry description and z is  
the index of the server associated with the emergency entry y. For each emergency  
entry (index y), one or more server entry (indexes (y,z)) can be configured. y and z  
must both follow single step increasing numbering starting at 1.  
Attribute  
Permitted Values  
Default  
Interpretation  
dialplan.x.routing.emer-  
gency.y.value  
Comma separated list  
of entries or single  
entry representing a or  
a combination of SIP  
URL.  
Null  
This represents the  
URLs that should be  
watched for emergency  
routing.  
Example:  
“15,17,18”,  
“911”, “sos”.  
When one of these  
defined URL is detected  
as being dialed by the  
user, the call will be  
automatically directed to  
the defined emergency  
server.  
dialplan.x.routing.emer-  
gency.y.server.z  
positive integer  
Null  
Index representing the  
server defined in  
4.6.2.4.2.1 Server  
<server/> on page 158  
that will be used for  
emergency routing.  
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4.6.2.5 Messaging <msg/>  
Message-waiting indication is supported on a per-registration basis.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
msg.bypassInstantMessage  
0, 1  
0
If set to 1, the display offering a choice  
of “Message Center” and “Instant Mes-  
sages” will be bypassed when pressing  
the Messages key. The phone will act as  
if “Message Center” was chosen. Refer  
to 3.6.1 Voice Mail Integration on  
page 64. Instant Messages will still be  
accessible from the Main Menu.  
4.6.2.5.1 Message Waiting Indicator <mwi/>  
In the following table, x is the registration number. IP 300, 301, and 430: x=1-2; IP  
500 and 501: x=1-3; IP 600: x=1-6; IP 601: x=1-12; IP 4000: x=1.  
Attribute  
Permitted Values  
Default  
Interpretation  
msg.mwi.x.sub-  
scribe  
ASCII encoded string con-  
taining digits (the user part  
of a SIP URL) or a string  
that constitutes a valid SIP  
URL (6416 or 6416@poly-  
com.com)  
Null  
If non-Null, the phone will  
send a SUBSCRIBE  
request to this contact after  
boot-up.  
msg.mwi.x.call-  
BackMode  
contact or  
registration or  
disabled  
“disabled”  
If set to “contact”, a call  
will be placed to the contact  
specified in the callback  
attribute when the user  
invokes message retrieval.  
If set to “registration”, a  
call will be placed using  
this registration to the con-  
tact registered (the phone  
will call itself).  
If set to “disabled”, mes-  
sage retrieval is disabled.  
msg.mwi.x.callBack ASCII encoded string con-  
taining digits (the user part  
of a SIP URL) or a string  
Null  
Contact to call when  
retrieving messages for this  
registration.  
that constitutes a valid SIP  
URL (6416 or 6416@poly-  
com.com)  
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4.6.2.6 Network Address Translation <nat/>  
These parameters define port and IP address changes used in NAT traversal. The port  
changes will change the port used by the phone, while the IP entry simply changes the  
IP advertised in the SIP signaling. This allows the use of simple NAT devices that can  
redirect traffic, but do not allow for port mapping. For example, port 5432 on the NAT  
device can be sent to port 5432 on an internal device, but not port 1234.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
nat.ip  
dotted-deci- Null  
mal IP  
address  
IP address to advertise within SIP signaling -  
should match the external IP address used by the  
NAT device.  
nat.signalPort  
1024 to  
65535  
Null  
Null  
If non-Null, this port will be used by the phone for  
SIP signaling, overriding the value set for voIp-  
Prot.local.signalPort in sip.cfg.  
nat.mediaPortStart  
1024 to  
65535  
If non-Null, this attribute will be used to set the  
initially allocated RTP port, overriding the value  
set for tcpIpApp.port.rtp.mediaPortRangeStart in  
sip.cfg. Refer to 4.6.1.10.3.1 RTP <RTP/> on  
page 124.  
nat.keepalive.inter- 0 to 3600  
val  
Null  
If non-Null (or 0), the keepalive interval in sec-  
onds. This parameter is used to set the interval at  
which phones will send a keep-alive packet to the  
gateway/NAT device to keep the communication  
port open so that NAT can continue to function as  
setup initially.  
®
The Microsoft Office Live Communications  
Server 2005 keepalive feature will override this  
interval. If you want to deploy phones behind a  
NAT and connect them to Live Communications  
Server, the keepalive interval received from the  
Live Communications Server must be short  
enough to keep the NAT port open. Once the TCP  
connection is closed, the phones stop sending  
keep-alive packets.  
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4.6.2.7 Attendant <attendant/>  
These attributes are available on SoundPoint® IP 600 and 601 phones (with an  
attached Expansion Module) only.  
The Busy Lamp Field (BLF) / attendant console feature enhances support for a phone-  
based attendant console.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
attendant.uri  
string  
Null  
For attendant console / busy lamp field (BLF) fea-  
ture. This specifies the list SIP URI on the server.  
If this is just a user part, the URI is constructed  
with the server host name/IP.  
attendant.reg  
positive  
integer  
1
For attendant console / BLF feature. This is the  
index of the registration which will be used to  
send a SUBSCRIBE to the list SIP URI specified  
in attendant.uri. For example, attendant.reg = 2  
means the second registration will be used.  
4.6.2.8 Roaming Buddies <roaming_buddies/>  
®
This attribute is used in conjunction with Microsoft Office Live Communications  
Server 2005 only.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
roaming_buddies.re positive  
integer  
Null  
Specifies the line/registration number which has  
roaming buddies support enabled. If Null, roam-  
ing buddies is disabled. If value < 1, then value is  
replaced with 1.  
g
Warning: This parameter must be enabled (value  
®
< 0) if the call server is Microsoft Office Live  
Communications Server 2005.  
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4.6.2.9 Roaming Privacy <roaming_privacy/>  
®
This attribute is used in conjunction with Microsoft Office Live Communications  
Server 2005 only.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
roaming_privacy.re positive  
integer  
Null  
Specifies the line/registration number which has  
roaming privacy support enabled. If Null, roam-  
ing privacy is disabled. If value < 1, then value is  
replaced with 1.  
g
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Session Initiation Protocol (SIP)  
5 Session Initiation Protocol (SIP)  
5.1 Basic Protocols  
All the basic calling functionality described in the SIP specification is supported.  
Refer to 5.1.1 RFC and Internet Draft Support on page 163 for supported RFC’s and  
drafts. Transfer is included in the basic SIP support.  
5.1.1 RFC and Internet Draft Support  
ID  
Title  
RFC 2387  
RFC 3261  
RFC 3262  
The MIME Multipart / Related Content-type  
SIP: Session Initiation Protocol (replacement for RFC 2543)  
Reliability of Provisional Responses in the Session Initiation Pro-  
tocol (SIP)  
RFC 3263  
RFC 3264  
Session Initiation Protocol (SIP): Locating SIP Servers  
An Offer / Answer Model with the Session Description Protocol  
(SDP)  
RFC 3265  
RFC 3515  
Session Initiation Protocol (SIP) - Specific Event Notification  
The Session Initiation Protocol (SIP) Refer Method  
draft-ietf-sip-cc-transfer-05.txt SIP Call Control - Transfer  
RFC 3891  
The Session Initiation Protocol (SIP) “Replaces” Header  
5.1.2 Request Support  
Method  
REGISTER  
INVITE  
ACK  
Supported  
Yes  
Notes  
Yes  
Yes  
CANCEL  
BYE  
Yes  
Yes  
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Method  
Supported  
Yes  
Notes  
OPTIONS  
SUBSCRIBE  
NOTIFY  
REFER  
Yes  
Yes  
Yes  
PRACK  
Yes  
5.1.3 Header Support  
In the following table, a “Yes” in the Supported column means the header is sent and  
properly parsed.  
Header  
Supported  
Yes  
No  
Notes  
Accept  
Accept-Encoding  
Accept-Language  
Alert-Info  
No  
Yes  
Yes  
Yes  
No  
Allow  
Allow-Events  
Authentication-Info  
Authorization  
Call-ID  
Yes  
Yes  
Yes  
Yes  
No  
Call-Info  
Contact  
Content-Disposition  
Content-Encoding  
Content-Language  
Content-Length  
Content-Type  
CSeq  
No  
No  
Yes  
Yes  
Yes  
No  
Date  
Diversion  
Yes  
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Header  
Supported  
No  
Notes  
Error-Info  
Event  
Yes  
Yes  
Yes  
No  
Expires  
From  
In-Reply-To  
Max-Forwards  
Min-Expires  
Min-SE  
Yes  
No  
Yes  
No  
MIME-Version  
Organization  
P-Asserted-Identity  
P-Preferred-Identity  
Priority  
No  
Yes  
Yes  
No  
Proxy-Authenticate  
Proxy-Authorization  
Proxy-Require  
RAck  
Yes  
Yes  
No  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
No  
Record-Route  
Refer-To  
Referred-By  
Remote-Party-ID  
Replaces  
Reply-To  
Require  
Yes  
No  
Retry-After  
Route  
Yes  
Yes  
No  
RSeq  
Server  
Session-Expires  
Subject  
Yes  
No  
Subscription-State  
Yes  
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Header  
Supported  
Yes  
Notes  
Supported  
Timestamp  
To  
No  
Yes  
Unsupported  
User-Agent  
Via  
No  
Yes  
Yes  
Warning  
No  
WWW-Authenticate  
Yes  
5.1.4 Response Support  
In the following table, a “Yes” in the Supported column means the header is parsed.  
The phone may not actually generate the response.  
5.1.4.1 1xx Responses - Provisional  
Response  
Supported  
Yes  
Notes  
100 Trying  
180 Ringing  
Yes  
181 Call Is Being Forwarded  
182 Queued  
No  
No  
183 Session Progress  
Yes  
5.1.4.2 2xx Responses - Success  
Response  
Supported  
Yes  
Notes  
200 OK  
202 Accepted  
Yes  
In REFER transfer.  
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5.1.4.3 3xx Responses - Redirection  
Response  
Supported  
Notes  
300 Multiple Choices  
Yes  
301 Moved Permanently Yes  
302 Moved Temporarily  
305 Use Proxy  
Yes  
No  
No  
380 Alternative Service  
5.1.4.4 4xx Responses - Request Failure  
Note  
All 4xx responses for which the phone does not provide specific support will be treated the same as  
400 Bad Request.  
Response  
Supported  
Yes  
Yes  
No  
Notes  
400 Bad Request  
401 Unauthorized  
402 Payment Required  
403 Forbidden  
No  
404 Not Found  
Yes  
Yes  
No  
405 Method Not Allowed  
406 Not Acceptable  
407 Proxy Authentication Required  
408 Request Timeout  
410 Gone  
Yes  
No  
No  
413 Request Entity Too Large  
414 Request-URI Too Long  
415 Unsupported Media Type  
416 Unsupported URI Scheme  
No  
No  
Yes  
No  
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Response  
Supported  
Notes  
420 Bad Extension  
No  
No  
No  
Yes  
Yes  
Yes  
No  
Yes  
No  
Yes  
Yes  
Yes  
No  
No  
421 Extension Required  
423 Interval Too Brief  
480 Temporarily Unavailable  
481 Call/Transaction Does Not Exist  
482 Loop Detected  
483 Too Many Hops  
484 Address Incomplete  
485 Ambiguous  
486 Busy Here  
487 Request Terminated  
488 Not Acceptable Here  
491 Request Pending  
493 Undecipherable  
5.1.4.5 5xx Responses - Server Failure  
Response  
Supported  
Yes  
Notes  
500 Server Internal Error  
501 Not Implemented  
502 Bad Gateway  
Yes  
No  
503 Service Unavailable  
504 Server Time-out  
505 Version Not Supported  
513 Message Too Large  
No  
No  
No  
No  
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5.1.4.6 6xx Responses - Global Failure  
Response  
Supported  
No  
Notes  
600 Busy Everywhere  
603 Decline  
Yes  
604 Does Not Exist Anywhere No  
606 Not Acceptable No  
5.1.5 Hold Implementation  
The phone supports both currently accepted means of signaling hold. The first method,  
no longer recommended due in part to the RTCP problems associated with it, is to set  
the “c” destination addresses for the media streams in the SDP to zero, for example,  
c=0.0.0.0. The second, and preferred, method is to signal the media directions with the  
“a” SDP media attributes sendonly, recvonly, inactive or sendrecv. The hold signaling  
method used by the phone is configurable (for more information, refer to 4.6.1.1.4 SIP  
<SIP/> on page 88) but both methods are supported when signaled by the remote end  
point.  
5.1.6 Reliability of Provisional Responses  
The phone fully supports RFC 3262 - Reliability of Provisional Responses.  
5.1.7 Transfer  
The phone supports transfer using the REFER method specified in draft-ietf-sip-cc-  
transfer-05 and RFC 3515.  
5.1.8 Third Party Call Control  
The phone supports the delayed media negotiations (INVITE without SDP) associated  
with third party call control applications.  
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5.2 Protocol Extensions  
The phone supports the following SIP protocol extensions.  
5.2.1 RFC and Internet Draft Support  
ID  
Title  
RFC 1321  
RFC 3311  
RFC 3325  
RFC 3725  
The MD5 Message-Digest Algorithm  
The Session Initiation Protocol (SIP) UPDATE Method  
SIP Asserted Identity  
Best Current Practices for Third Party Call Control  
(3pcc) in the Session Initiation Protocol (SIP)  
RFC 3842  
A Message Summary and Message Waiting Indication  
Event Package for the Session Initiation Protocol (SIP)  
draft-anil-sipping-bla-02.txt  
draft-ietf-simple-event-list-07.txt  
Implementing Bridged Line Appearances (BLA)  
Using Session Initiation Protocol (SIP)  
Session Initiation Protocol (SIP) Event Notification  
Extension for Resource Lists  
draft-levy-sip-diversion-04.txt  
draft-ietf-sip-session-timer-12.txt  
Diversion Indication in SIP  
Session Timers in the Session Initiation Protocol (SIP)  
draft-ietf-sipping-dialog-package-06.txt INVITE Initiated Dialog Event Package for the Session  
Initiation Protocol (SIP)  
draft-ietf-sip-privacy-04.txt  
SIP Extensions for Network-Asserted Caller Identity and  
Privacy within Trusted Networks  
draft-ietf-sip-referredby-05.txt  
draft-levy-sip-diversion-06.txt  
SIP Referred by Mechanism  
Diversion Indication in SIP  
draft-ietf-sipping-cc-conferencing-  
03.txt  
SIP Call Control - Conferencing for User Agents  
draft-ietf-sip-connect-reuse-04  
Connection Reuse in the Session Initiation Protocol (SIP)  
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5.2.2 Request Support  
Method  
Supported  
Notes  
INFO  
Yes  
RFC 2976, the phone does not generate INFO requests, but will  
issue a final response upon receipt. No INFO message bodies  
are parsed.  
MESSAGE  
UPDATE  
Yes  
Yes  
Final response is sent upon receipt. Message bodies of type  
text/plain are sent and received.  
5.2.3 SIP for Instant Messaging and Presence Leverag-  
ing Extensions  
The phone is compatible with the Presence and Instant Messaging features of  
Microsoft® Windows® Messenger 5.1. In a future release, support for the Presence and  
Instant Message recommendations in the SIP Instant Messaging and Presence Lever-  
aging Extensions (SIMPLE) proposals will be provided:  
• draft-ietf-simple-cpim-mapping-01  
• draft-ietf-simple-presence-07  
• draft-ietf-simple-presencelist-package-00  
• draft-ietf-simple-winfo-format-02  
• draft-ietf-simple-winfo-package-02  
or their successors.  
5.2.4 Shared Call Appearance Signaling  
A shared line is an address of record managed by a server. The server allows multiple  
end points to register locations against the address of record.  
The phone supports shared call appearances (SCA) using the SUBSCRIBE-NOTIFY  
method in the “SIP Specific Event Notification” framework (RFC 3265). The events  
used are:  
• “call-info” for call appearance state notification  
• “line-seize for the phone to ask to seize the line  
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5.2.5 Bridged Line Appearance Signaling  
A bridged line is an address of record managed by a server. The server allows multiple  
end points to register locations against the address of record.  
The phone supports bridged line appearances (BLA) using the SUBSCRIBE-NOTIFY  
method in the “SIP Specific Event Notification” framework (RFC 3265). The events  
used are:  
• “dialog” for bridged line appearance subscribe and notify  
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Appendix 1  
6 Appendix 1  
6.1 Trusted Certificate Authority List  
The following certificate authorities are trusted by the phone by default.  
ABAecom (sub., Am. Bankers Assn.) Root CA  
ANX Network CA by DST  
American Express CA  
American Express Global CA  
BelSign Object Publishing CA  
BelSign Secure Server CA  
Deutsche Telekom AG Root CA  
Digital Signature Trust Co. Global CA 1  
Digital Signature Trust Co. Global CA 2  
Digital Signature Trust Co. Global CA 3  
Digital Signature Trust Co. Global CA 4  
Entrust Worldwide by DST  
Entrust.net Premium 2048 Secure Server CA  
Entrust.net Secure Personal CA  
Entrust.net Secure Server CA  
Equifax Premium CA  
Equifax Secure CA  
GTE CyberTrust Global Root  
GTE CyberTrust Japan Root CA  
GTE CyberTrust Japan Secure Server CA  
GTE CyberTrust Root 2  
GTE CyberTrust Root 3  
GTE CyberTrust Root 4  
GTE CyberTrust Root 5  
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GTE CyberTrust Root CA  
GlobalSign Partners CA  
GlobalSign Primary Class 1 CA  
GlobalSign Primary Class 2 CA  
GlobalSign Primary Class 3 CA  
GlobalSign Root CA  
National Retail Federation by DST  
TC TrustCenter, Germany, Class 1 CA  
TC TrustCenter, Germany, Class 2 CA  
TC TrustCenter, Germany, Class 3 CA  
TC TrustCenter, Germany, Class 4 CA  
Thawte Personal Basic CA  
Thawte Personal Freemail CA  
Thawte Personal Premium CA  
Thawte Premium Server CA  
Thawte Server CA  
Thawte Universal CA Root  
UPS Document Exchange by DST  
ValiCert Class 1 VA  
ValiCert Class 2 VA  
ValiCert Class 3 VA  
VeriSign Class 4 Primary CA  
Verisign Class 1 Public Primary Certification Authority  
Verisign Class 1 Public Primary Certification Authority - G2  
Verisign Class 1 Public Primary Certification Authority - G3  
Verisign Class 2 Public Primary Certification Authority  
Verisign Class 2 Public Primary Certification Authority - G2  
Verisign Class 2 Public Primary Certification Authority - G3  
Verisign Class 3 Public Primary Certification Authority  
Verisign Class 3 Public Primary Certification Authority - G2  
Verisign Class 3 Public Primary Certification Authority - G3  
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Verisign Class 4 Public Primary Certification Authority - G2  
Verisign Class 4 Public Primary Certification Authority - G3  
Verisign/RSA Commercial CA  
Verisign/RSA Secure Server CA  
6.2 Miscellaneous Administrative Tasks  
6.2.1 Adding a Background Logo  
This section provides instructions on how to add a background logo to all Sound-  
®
Point IP phones in your organization. You must be running at least BootROM 2.x.x  
®
and SIP 1.x.x. One bitmap file is required for each model, but SoundPoint IP 301  
phones do not support bitmap logos.  
Model  
Width  
n/a  
Height  
n/a  
Color Depth  
IP 300/301  
IP 430  
n/a  
94  
23  
monochrome  
IP 500/501  
IP 600/601  
IP 4000  
114  
209  
150  
51  
4-bit grayscale or monochrome  
4-bit grayscale or monochrome  
4-bit grayscale or monochrome  
109  
33  
Logos smaller than described in the table above are acceptable, but larger logos may  
be truncated or interfere with other areas of the user interface.  
®
The SoundPoint IP 500/501/600/601 phones only support the four colors listed  
below. Any other colors will be approximated.  
®
The SoundPoint IP 4000 phone only supports black and white. Any other colors will  
be rendered as either black or white.  
RGB Values  
(Decimal)  
RGB Values  
(Hexadecimal)  
Color  
Black  
0,0,0  
00,00,00  
60,60,60  
A0,A0,A0  
FF,FF,FF  
Dark Gray  
Light Gray  
White  
96,96,96  
160,160,160  
255,255,255  
Copyright © 2006 Polycom, Inc.  
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Configuration File Changes  
In the <bitmaps> section of sip.cfg, find the end of each model's bitmap list and add  
your bitmap to the end; do not include the .bmp extension:  
<bitmaps>  
<IP_300 … />  
<IP_500 … bitmap.IP_500.66.name="logo-500" />  
<IP_600 … bitmap.IP_600.70.name="logo-600" />  
<IP_4000 … bitmap.IP_4000.70.name="logo-4000" />  
</bitmaps>  
Next, enable the idle display feature and modify the IDLE_DISPLAY "animation" for  
each model to point to your bitmap (again without the .bmp extension):  
<indicators ind.idleDisplay.enabled="1">  
<Animations>  
<IP_300>  
…
</IP_300>  
<IP_500>  
…
<IDLE_DISPLAY ind.anim.IP_500.38.frame.1.bitmap="logo-500"  
ind.anim.IP_500.38.frame.1.duration="0"/>  
…
</IP_500>  
<IP_600>  
…
<IDLE_DISPLAY ind.anim.IP_600.38.frame.1.bitmap="logo-600"  
ind.anim.IP_600.38.frame.1.duration="0"/>  
…
</IP_600>  
<IP_4000>  
…
<IDLE_DISPLAY ind.anim.IP_4000.38.frame.1.bitmap="logo-4000"  
ind.anim.IP_4000.38.frame.1.duration="0"/>  
…
</IP_4000>  
</Animations>  
…
</indicators>  
Finally, edit the {MAC}.cfg file to instruct the phone to download the bitmap files at  
boot time:  
MISC_FILES="logo-500.bmp" [for SPIP 500/501 phones]  
MISC_FILES="logo-600.bmp" [for SPIP 600/601 phones]  
MISC_FILES="logo-4000.bmp" [for SSIP 4000 phones]  
Many configuration-generation systems do not make it easy to customize the contents  
of this file based on the model; if you are using one of these systems, you can have all  
phones download all the bitmaps:  
MISC_FILES="logo-500.bmp, logo-600.bmp, logo-4000.bmp" [for all phones]  
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Appendix 2  
7 Appendix 2  
7.1 Third Party Software Attribution  
The following third party software products are part of the Session Initiation Protocol  
(SIP) application.  
Ares:  
Copyright 1998 by the Massachusetts Institute of Technology.  
Permission to use, copy, modify, and distribute this software and its documentation for  
any purpose and without fee is hereby granted, provided that the above copyright  
notice appear in all copies and that both that copyright notice and this permission  
notice appear in supporting documentation, and that the name of M.I.T. not be used in  
advertising or publicity pertaining to distribution of the software without specific,  
written prior permission.  
M.I.T. makes no representations about the suitability of this software for any purpose.  
It is provided "as is" without express or implied warranty.  
OpenSSL  
The OpenSSL toolkit stays under a dual license, i.e. both the conditions of the  
OpenSSL License and the original SSLeay license apply to the toolkit. See below for  
the actual license texts. Actually both licenses are BSD-style Open Source licenses. In  
case of any license issues related to OpenSSL please contact openssl-  
[email protected].  
OpenSSL License  
Copyright (c) 1998-2003 The OpenSSL Project. All rights reserved.  
Redistribution and use in source and binary forms, with or without modification, are  
permitted provided that the following conditions are met:  
1. Redistributions of source code must retain the above copyright notice, this list of  
conditions and the following disclaimer.  
2. Redistributions in binary form must reproduce the above copyright notice, this list  
of conditions and the following disclaimer in the documentation and/or other materials  
provided with the distribution.  
3. All advertising materials mentioning features or use of this software must display  
the following acknowledgment:  
"This product includes software developed by the OpenSSL Project for use in the  
OpenSSL Toolkit. (http://www.openssl.org/)"  
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4. The names "OpenSSL Toolkit" and "OpenSSL Project" must not be used to endorse  
or promote products derived from this software without prior written permission. For  
written permission, please contact [email protected].  
5. Products derived from this software may not be called "OpenSSL" nor may  
"OpenSSL" appear in their names without prior written permission of the OpenSSL  
Project.  
6. Redistributions of any form whatsoever must retain the following acknowledgment:  
"This product includes software developed by the OpenSSL Project for use in the  
OpenSSL Toolkit (http://www.openssl.org/)"  
THIS SOFTWARE IS PROVIDED BY THE OpenSSL PROJECT ``AS IS'' AND  
ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIM-  
ITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FIT-  
NESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT  
SHALL THE OpenSSL PROJECT OR ITS CONTRIBUTORS BE LIABLE FOR  
ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSE-  
QUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCURE-  
MENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR  
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON  
ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABIL-  
ITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN  
ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF  
THE POSSIBILITY OF SUCH DAMAGE.  
This product includes cryptographic software written by Eric Young (eay@crypt-  
soft.com). This product includes software written by Tim Hudson (tjh@crypt-  
soft.com).  
Original SSLeay License:  
Copyright (C) 1995-1998 Eric Young ([email protected])  
All rights reserved.  
This package is an SSL implementation written by Eric Young ([email protected]).  
The implementation was written so as to conform with Netscape’s SSL.  
This library is free for commercial and non-commercial use as long as the following  
conditions are adhered to. The following conditions apply to all code found in this dis-  
tribution, be it the RC4, RSA, lhash, DES, etc., code; not just the SSL code. The SSL  
documentation included with this distribution is covered by the same copyright terms  
except that the holder is Tim Hudson ([email protected]).  
Copyright remains Eric Young's, and as such any Copyright notices in the code are not  
to be removed. If this package is used in a product, Eric Young should be given attri-  
bution as the author of the parts of the library used. This can be in the form of a textual  
message at program startup or in documentation (online or textual) provided with the  
package. Redistribution and use in source and binary forms, with or without modifica-  
tion, are permitted provided that the following conditions are met:  
1. Redistributions of source code must retain the copyright notice, this list of condi-  
tions and the following disclaimer.  
2. Redistributions in binary form must reproduce the above copyright notice, this list  
of conditions and the following disclaimer in the documentation and/or other materials  
provided with the distribution.  
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3. All advertising materials mentioning features or use of this software must display  
the following acknowledgement: "This product includes cryptographic software writ-  
ten by Eric Young ([email protected])"  
The word 'cryptographic' can be left out if the routines from the library being used are  
not cryptographic related.  
4. If you include any Windows specific code (or a derivative thereof) from the apps  
directory (application code) you must include an acknowledgement: "This product  
includes software written by Tim Hudson ([email protected])"  
THIS SOFTWARE IS PROVIDED BY ERIC YOUNG ``AS IS'' AND ANY  
EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,  
THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A  
PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE  
AUTHOR OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,  
INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES  
(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE  
GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS  
INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABIL-  
ITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING  
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF  
THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAM-  
AGE.  
The licence and distribution terms for any publicly available version or derivative of  
this code cannot be changed. i.e. this code cannot simply be copied and put under  
another distribution licence [including the GNU Public Licence.]  
zlib:  
(C) 1995-2002 Jean-loup Gailly and Mark Adler  
This software is provided 'as-is', without any express or implied warranty. In no event  
will the authors be held liable for any damages arising from the use of this software.  
Permission is granted to anyone to use this software for any purpose, including com-  
mercial applications, and to alter it and redistribute it freely, subject to the following  
restrictions:  
1. The origin of this software must not be misrepresented; you must not claim that you  
wrote the original software. If you use this software in a product, an acknowledgment  
in the product documentation would be appreciated but is not required.  
2. Altered source versions must be plainly marked as such, and must not be misrepre-  
sented as being the original software.  
3. This notice may not be removed or altered from any source distribution.  
Jean-loup Gailly  
[email protected]  
Mark Adler  
[email protected]  
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Appendix 2  
Expat:  
Copyright (c) 1998, 1999, 2000 Thai Open Source Software Center Ltd and Clark  
Cooper  
Permission is hereby granted, free of charge, to any person obtaining a copy of this  
software and associated documentation files (the "Software"), to deal in the Software  
without restriction, including without limitation the rights to use, copy, modify, merge,  
publish, distribute, sublicense, and/or sell copies of the Software, and to permit per-  
sons to whom the Software is furnished to do so, subject to the following conditions:  
The above copyright notice and this permission notice shall be included in all copies  
or substantial portions of the Software.  
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY  
KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE  
WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PUR-  
POSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR  
COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER  
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHER-  
WISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE  
OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.  
curl:  
COPYRIGHT AND PERMISSION NOTICE  
Copyright (c) 1996 - 2004, Daniel Stenberg, <[email protected]>.  
All rights reserved.  
Permission to use, copy, modify, and distribute this software for any purpose with or  
without fee is hereby granted, provided that the above copyright notice and this per-  
mission notice appear in all copies.  
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY  
KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE  
WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PUR-  
POSE AND NONINFRINGEMENT OF THIRD PARTY RIGHTS. IN NO EVENT  
SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY  
CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF  
CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CON-  
NECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN  
THE SOFTWARE.  
Except as contained in this notice, the name of a copyright holder shall not be used in  
advertising or otherwise to promote the sale, use or other dealings in this Software  
without prior written authorization of the copyright holder.  
180  
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