Polycom Network Router Version 14x 17 User Manual
Administrator Guide  
®
®
SoundPoint /SoundStation IP SIP  
Version 1.4.x  
17 November 2004  
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Administrator Guide - SoundPoint® IP / SoundStation® IP  
Table of Contents  
Table of Contents  
1 Overview......................................................... 1  
2 Installation and Operation ................................. 3  
2.1 Installation Models..................................................3  
2.2 Installation Process..................................................4  
2.2.1 Basic Network Setup....................................................................................................................... 4  
2.2.1.1 DHCP or Manual TCP/IP Setup..............................................................................................5  
2.2.1.2 Local User Interface Setup Menus ..........................................................................................6  
2.2.1.3 Reset to Factory Defaults......................................................................................................9  
2.2.2 Application Configuration................................................................................................................ 9  
2.2.2.1 Centralized Configuration......................................................................................................10  
2.2.2.2 Local Phone Configuration.....................................................................................................15  
3 Features .......................................................... 17  
3.1 Basic Features.........................................................17  
3.1.1 Call Log ......................................................................................................................................... 17  
3.1.2 Call Timer ...................................................................................................................................... 17  
3.1.3 Call Waiting ................................................................................................................................... 18  
3.1.4 Called Party Identification............................................................................................................... 18  
3.1.5 Calling Party Identification.............................................................................................................. 18  
3.1.6 Missed Call Notification................................................................................................................... 18  
3.1.7 Configurable Feature Keys.............................................................................................................. 19  
3.1.8 Connected Party Identification ........................................................................................................ 23  
3.1.9 Context Sensitive Volume Control.................................................................................................... 23  
3.1.10 Customizable Audio Sound Effects................................................................................................. 23  
3.1.11 Message Waiting Indication .......................................................................................................... 24  
3.1.12 Distinctive Incoming Call Treatment............................................................................................... 24  
3.1.13 Distinctive Ringing........................................................................................................................ 25  
3.1.13 Distinctive Call Waiting ................................................................................................................. 25  
3.1.14 Do-Not-Disturb............................................................................................................................. 26  
3.1.15 Handset, Headset, and Speakerphone........................................................................................... 26  
3.1.16 Local Contact Directory................................................................................................................. 27  
3.1.16.1 Local Contact Directory File Format.......................................................................................28  
3.1.17 Local Digit Map ............................................................................................................................ 30  
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3.1.18 Microphone Mute ......................................................................................................................... 30  
3.1.19 Multiple Call Appearances............................................................................................................. 30  
3.1.20 Shared Call Appearances .............................................................................................................. 31  
3.1.21 Bridged Line Appearances............................................................................................................. 32  
3.1.22 Customizable Fonts and Indicators................................................................................................. 33  
3.1.23 Soft Key-Driven User Interface...................................................................................................... 34  
3.1.24 Speed Dial ................................................................................................................................... 34  
3.1.25 Time and Date Display.................................................................................................................. 35  
3.1.26 Idle Display Animation ................................................................................................................. 36  
3.2 Call Management Features........................................37  
3.2.1 Automatic Off-hook Call Placement................................................................................................. 37  
3.2.2 Call Hold........................................................................................................................................ 37  
3.2.3 Call Transfer .................................................................................................................................. 38  
3.2.4 Three-Way Conference, Local or Centralized.................................................................................... 38  
3.2.5 Call Diversion (Call Forward) .......................................................................................................... 38  
3.2.6 Directed Call Pick-up ...................................................................................................................... 39  
3.2.7 Group Call Pick-up.......................................................................................................................... 40  
3.2.8 Call Park / Retrieve ....................................................................................................................... 40  
3.2.9 Last Call Return.............................................................................................................................. 41  
3.3 Audio Processing Features ........................................41  
3.3.1 Low-Delay Audio Packet Transmission ............................................................................................. 41  
3.3.2 Jitter Buffer and Packet Error Concealment ..................................................................................... 41  
3.3.3 Local Conference Mixing................................................................................................................. 42  
3.3.4 Voice Activity Detection (VAD)......................................................................................................... 42  
3.3.5 DTMF Tone Generation ................................................................................................................... 43  
3.3.6 DTMF Event RTP Payload ................................................................................................................ 43  
3.3.7 Acoustic Echo Cancellation (AEC) ..................................................................................................... 44  
3.3.8 Audio Codecs.................................................................................................................................. 44  
3.3.9 Background Noise Suppression (BNS).............................................................................................. 45  
3.3.10 Comfort Noise Fill......................................................................................................................... 45  
3.3.11 Automatic Gain Control (AGC)....................................................................................................... 45  
3.4 Presence and Instant Messaging Features ...................45  
3.4.1 Presence........................................................................................................................................ 45  
3.4.2 Instant Messaging .......................................................................................................................... 46  
3.5 Localization Features ...............................................46  
3.5.1 Multilingual User Interface ............................................................................................................. 46  
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3.5.2 Downloadable Fonts ....................................................................................................................... 48  
3.5.3 Synthesized Call Progress Tones...................................................................................................... 48  
3.6 Advanced Server Features ........................................49  
3.6.1 Voicemail Integration ..................................................................................................................... 49  
3.6.2 Multiple Registrations..................................................................................................................... 50  
3.6.3 ACD login / logout ......................................................................................................................... 52  
3.6.4 ACD agent available / unavailable.................................................................................................. 53  
3.6.5 Server Redundancy ........................................................................................................................ 53  
3.6.5.1 DNS SIP Server Name Resolution...........................................................................................54  
3.7 Accessory Internet Features.......................................54  
3.7.1 MicroBrowser................................................................................................................................. 54  
3.8 Security Features.....................................................54  
3.8.1 Local User and Administrator Privilege Levels.................................................................................. 54  
3.8.2 Incoming Signaling Validation......................................................................................................... 55  
4 Optimization .................................................... 57  
4.1 Ethernet Switch .......................................................57  
4.2 Application Network Setup .......................................57  
4.2.1 RTP Ports ....................................................................................................................................... 57  
4.2.2 Working with Network Address Translation (NAT) ............................................................................ 58  
4.3 Updating and Rebooting...........................................59  
4.4 Event Logging.........................................................60  
4.5 Audio Quality Issues and VLANs ................................61  
4.5.1 IP TOS ........................................................................................................................................... 62  
4.5.2 IEEE 802.1p/Q............................................................................................................................... 62  
4.5.3 RTCP Support ................................................................................................................................. 63  
4.6 Configuration Files...................................................65  
4.6.1 Core Configuration - ipmid.cfg ........................................................................................................ 65  
4.6.1.1 Localization <localization/> .................................................................................................65  
4.6.1.2 User Preferences <user_preferences/> ..................................................................................69  
4.6.1.3 Tones <tones/> .................................................................................................................70  
4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> ..............................................................72  
4.6.1.5 Sound Effects <sound_effects/>...........................................................................................73  
4.6.1.6 Voice Settings <voice/> ......................................................................................................78  
4.6.1.7 Quality of Service <QOS/> ..................................................................................................86  
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Table of Contents  
4.6.1.8 Basic TCP/IP <TCP_IP/>.....................................................................................................88  
4.6.1.9 Web Server <HTTPD/>........................................................................................................91  
4.6.1.10 Call Handling Configuration <call/>.....................................................................................92  
4.6.1.11 Directory <directory/>.......................................................................................................93  
4.6.1.12 Fonts <font/>..................................................................................................................94  
4.6.1.13 Keys <keys/>..................................................................................................................96  
4.6.1.14 Bitmaps <bitmaps/>.........................................................................................................100  
4.6.1.15 Indicators <indicators/>.....................................................................................................100  
4.6.1.16 Event Logging <logging/> .................................................................................................103  
4.6.1.17 Security <security/> .........................................................................................................106  
4.6.1.18 Provisioning <provisioning/>..............................................................................................107  
4.6.1.19 RAM Disk <RAMdisk/>......................................................................................................107  
4.6.1.20 Request <request/>..........................................................................................................108  
4.6.1.21 Feature <feature/>...........................................................................................................109  
4.6.1.22 Resource <resource/>.......................................................................................................110  
4.6.1.23 MicroBrowser <microbrowser/>..........................................................................................111  
4.6.2 SIP Configuration - sip.cfg .............................................................................................................. 113  
4.6.2.1 Protocol <volpProt/>...........................................................................................................113  
4.6.3 Per-phone Configuration - phone1.cfg............................................................................................. 121  
4.6.3.1 Registration <reg/> ............................................................................................................121  
4.6.3.2 Calls <call/>......................................................................................................................123  
4.6.3.3 Diversion <divert/>.............................................................................................................124  
4.6.3.4 Dial Plan <dialplan/>..........................................................................................................126  
4.6.3.5 Messaging <msg/> ............................................................................................................128  
4.6.3.6 Network Address Translation <nat/> .....................................................................................129  
5 Session Initiation Protocol (SIP)........................... 131  
5.1 Basic Protocols........................................................131  
5.1.1 RFC and Internet Draft Support....................................................................................................... 131  
5.1.2 Request Support............................................................................................................................. 131  
5.1.3 Header Support.............................................................................................................................. 132  
5.1.4 Response Support........................................................................................................................... 134  
5.1.4.1 1xx Responses - Provisional ..................................................................................................134  
5.1.4.2 2xx Responses - Success ......................................................................................................134  
5.1.4.3 3xx Responses - Redirection..................................................................................................135  
5.1.4.4 4xx Responses - Request Failure............................................................................................135  
5.1.4.5 5xx Responses - Server Failure ..............................................................................................136  
5.1.4.6 6xx Responses - Global Failure ..............................................................................................136  
5.1.5 Hold Implementation...................................................................................................................... 137  
5.1.6 Reliability of Provisional Responses................................................................................................. 137  
5.1.7 Transfer......................................................................................................................................... 137  
5.1.8 Third Party Call Control................................................................................................................... 137  
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Table of Contents  
5.2 Protocol Extensions..................................................138  
5.2.1 RFC and Internet Draft Support....................................................................................................... 138  
5.2.2 Request Support............................................................................................................................. 138  
5.2.3 SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE).......................................... 139  
5.2.4 Shared Call Appearance Signaling................................................................................................... 139  
5.2.5 Bridged Line Appearance Signaling ................................................................................................. 139  
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Table of Contents  
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Administrator Guide - SoundPoint® IP / SoundStation® IP  
Overview  
1 Overview  
This Administrator Guide is for the SIP 1.4.0 software release, and the bootROM 2.6.0  
release.  
SoundPoint® 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 and text message communications. These phones are endpoints in  
the overall network topology designed to interoperate with other compatible equip-  
ment including application servers, media servers, internetworking gateways, voice  
bridges, and other endpoints.  
Remote  
Boot Server  
PSTN  
Internet  
IDC  
Remote  
Application  
Server  
Router /  
Firewall  
PSTN  
Gateway  
M
odem Bank  
7x  
8x  
9x  
1
0
x
1
1
x
1
2
x
7x  
8x  
9x  
1
0
x
1
1
x
12x  
C
A
7
1
8
9
1
0
1
1
12  
Ethernet  
Switches  
5
6
1x  
7x  
2
x
3x  
9x  
4
x
5
x
6
x
1
x
2
x
3
x
4x  
B
5x  
6x  
A
8x  
1
0
x
1
1
x
1
2
x
7x  
8x  
9x  
1
0
x
1
1
x
12x  
C
A
7
1
8
2
9
3
1
01  
112  
4
5
6
1x  
2x  
3x  
4x  
5x  
6x  
1x  
2x  
3x  
4x  
B
5x  
6x  
A
10/100  
Ethernet  
Switch  
7x  
1x  
8
x
9
x
1
0
x
1
1
x
1
2
x
7x  
1x  
8x  
2x  
9
x
10  
x
11  
x
12x  
6x  
C
A
7
1
8
2
9
1
0
1
1
12  
4
6
2x  
3x  
4x  
5
x
6x  
3x  
4x  
5x  
A
B
Voice Bridge  
Polycom  
SoundPoint IP  
500/600s  
IDC  
PC  
PC  
Local  
Application  
Server  
Local  
Boot Server  
10/100  
Ethernet  
Hub  
7x  
8x  
9x  
1
0x  
1
1
x
1
2
x
7
x
8
x
9
x
10  
x
11x  
12x  
C
7
8
9
1
0
1
1
12  
A
1
3
4
1x  
2x  
3x  
4
x
5
x
6x  
1x  
2x  
3x  
4x  
5x  
6x  
A
B
PC  
PC  
The phones connect physically to a standard office twisted-pair (IEEE 802.3) 10/100  
megabytes per second Ethernet LAN and send and receive all data using the same  
packet-based technology. Since the phone is a data terminal, digitized audio being just  
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Administrator Guide - SoundPoint® IP / SoundStation® IP  
Overview  
another type of data from its perspective, the phone is capable of vastly more than tra-  
ditional business phones. As SoundPoint® IP and SoundStation® IP run the same 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 is fundamentally a good office phone, providing the productiv-  
ity enhancing features needed today such as multiple call appearances, full-duplex  
speakerphone, hold, transfer, conference, forward, voice mail compatibility, and con-  
tact directory.  
2
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Administrator Guide - SoundPoint® IP / SoundStation® IP  
Installation and Operation  
2 Installation and Operation  
This section describes the basic steps that are needed to make your phone operational.  
2.1 Installation Models  
There are diverse installation models scaling from stand-alone phones to large, cen-  
trally provisioned systems with thousands of phones. For any size system, the phones  
can be centrally provisioned from a boot server via a system of global and per-phone  
configuration files. To augment the central provisioning model, or as the sole method  
in smaller systems, configuration can be done using user interfaces driven from the  
phones themselves: both a local setup user interface and a web server-based user inter-  
face are available to make configuration changes.  
Boot Server  
config overrides  
directory  
master config file  
application binary  
config files  
dictionary files  
user interface  
resource files  
event log files  
0004f2002999.cfg  
sip.ld  
ipmid.cfg  
sip.cfg  
phone2999.cfg  
SoundPointIP-dictionary.xml  
0004f2002999-phone.cfg  
0004f2002999-directory.cfg  
0004f2002999-boot.log  
0004f2002999-app.log  
SoundPoint IP  
Local User Interface  
MAC 00:04:f2:00:29:99  
Local  
Web Server  
A boot server allows global and per-phone configuration to be managed centrally via  
text XML-format configuration files that are downloaded by the phones at boot time.  
The boot server also facilitates automated application upgrades, diagnostics, and a  
measure of fault tolerance.  
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Administrator Guide - SoundPoint® IP / SoundStation® IP  
Installation and Operation  
The configuration served by the boot server can be augmented by changes made  
locally on the phone itself or via the phone’s built-in web server. If file uploads are  
permitted, the boot server allows these local changes to be backed up automatically.  
Polycom recommends the boot server central provisioning model for installations  
involving more than a few phones. The investment required is minimal in terms of  
time and equipment, and the benefits are significant.  
The phones also support dynamic host configuration protocol (DHCP). When set up,  
DHCP permits plug-and-play TCP/IP network setup.  
2.2 Installation Process  
Regardless of whether or not you will be installing a centrally provisioned system,  
there are two steps required to get your phones up and running.  
1. Basic TCP/IP Network Setup such as IP address and subnet mask. For more infor-  
mation, see 2.2.1 Basic Network Setup on page 4.  
2. Application Configuration such as application specific parameters. For  
more information, see 2.2.2 Application Configuration on page 9.  
2.2.1 Basic Network Setup  
The phones boot up in two phases:  
• Phase 1: bootROM - a generic program designed to load the application.  
• Phase 2: application - the SIP phone application.  
Networking starts in Phase 1. The bootROM application uses the network to query the  
boot server for upgrades or configuration changes, which is an optional process that  
will happen automatically when properly deployed. The boot server can be on the  
local LAN or anywhere on the Internet. The bootROM then loads the configured  
application. The application will restart networking using most of the parameters  
established by the bootROM (a DHCP query will be performed by the application).  
Basic network settings can be changed during Phase 1 using the bootROM’s setup  
menu. A similar, but more sophisticated menu system is present in the application for  
changing the same network parameters. For more information, see 2.2.1.2 Local User  
Interface Setup Menus on page 6.  
4
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Administrator Guide - SoundPoint® IP / SoundStation® IP  
Installation and Operation  
2.2.1.1 DHCP or Manual TCP/IP Setup  
Basic network settings can be derived from DHCP or entered manually using the  
phone’s LCD-based user interface. Polycom recommends using DHCP where possi-  
ble 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:  
Configuration File  
(Phase 2: application only)  
Local  
FLASH  
Parameter  
DHCP Option  
DHCP  
ꢀpriority when more than one source exists ꢀ  
1
2
3
1
1
3
IP address  
•
•
•
•
-
-
-
-
•
•
•
•
subnet mask  
IP gateway  
See 2.2.1.2.2  
DHCP Menu  
on page 7  
boot server address  
42 then 4  
SNTP server address  
SNTP GMT offset  
•
•
•
•
•
•
-
-
•
•
•
•
2
6
6
DNS server IP address  
alternate DNS server IP  
address  
15  
DNS domain  
•
-
•
a
See 2.2.1.2.2  
DHCP Menu  
on page 7  
Special Case: Cisco Discovery Protocol (CDP) over-  
rides Local FLASH which overrides DHCP VLAN  
Discovery.  
VLAN ID  
a. Can be obtained from a connected Ethernet switch if the switch supports CDP.  
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Installation and Operation  
2.2.1.2 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>Network Configuration. This menu is  
locked by default. Enter the administrator password to unlock.  
(Factory default password: 456)  
Phone network configuration parameters may be edited by means of a main menu and  
two sub-menus: DHCP Menu and Server Menu.  
Use the soft keys, the arrow keys, the Sel/3, and the Del/X keys to make changes.  
Parameters that cannot be changed are read-only due to the value of other parameters.  
For example, if the DHCP Client parameter is enabled, the Phone IP Addr and Subnet  
Mask parameters are dimmed or not visible since these are guaranteed to be supplied  
by the DHCP server (mandatory DHCP parameters) and the statically assigned IP  
address and subnet mask will never be used in this configuration.  
2.2.1.2.1 Main Menu  
Configuration parameters that may be edited on the main setup menu are described in  
the table below:  
Name  
Possible 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.  
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.  
IP Gateway  
dotted-decimal IP  
address  
Phone’s default router.  
SNTP Address  
dotted-decimal IP  
address  
SNTP server from which the phone will obtain  
the current time.  
OR  
domain name string  
6
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Installation and Operation  
Name  
Possible Valuesa  
Description  
GMT Offset  
-12 through +13  
Offset of the local time zone from Greenwich  
Mean Time in half hour increments.  
DNS Server  
dotted-decimal IP  
address  
Primary server to which the phone directs  
Domain Name System queries.  
DNS Alternate Server dotted-decimal IP  
address  
Secondary server to which the phone directs  
Domain Name System queries.  
DNS Domain  
CDP  
domain name string  
Enabled, Disabled  
Phone’s DNS domain.  
If enabled, the phone will attempt to determine  
its VLAN ID via the CDP.  
VLAN ID  
Null, 0 through 4095  
Phone’s 802.1Q VLAN identifier.  
Note: 4095 = no VLAN tagging  
a. A parameter value of “???” indicates that the parameter has not yet been set and saved in the  
phone’s configuration. Any such parameter should have its value set before continuing.  
The DHCP and Server sub-menus may be accessed from the main setup menu.  
2.2.1.2.2 DHCP Menu  
The DHCP menu is accessible only when the DHCP client is enabled. DHCP config-  
uration 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.  
Boot Server  
Option 66  
Custom  
Option 66: The phone will look for option number 66  
(string type) in the response received from the DHCP  
server. The DHCP server must be configured to send  
the boot server address in option 66.  
Custom: The phone will look for the option number  
specified by the “Boot Server Option” parameter  
(below), and the type specified by the “Boot Server  
Option Type” parameter (below) in the response  
received from the DHCP server.  
Static  
Static: The phone will use the boot server configured  
via the Server Menu. For more information, see  
2.2.1.2.3 Server Menu on page 8.  
Custom+Opt.66 Custom+Opt.66: The phone will first use the custom  
option if present or use Option 66 if the custom option  
is not present.  
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Installation and Operation  
Possible  
Values  
Name  
Description  
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,  
String  
When the Boot Server parameter is set to Custom, this  
parameter specifies the type of the DHCP option in  
which the phone will look for its boot server.  
VLAN Discovery  
Disabled  
Fixed  
No VLAN discovery via 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). Default is 129.  
2.2.1.2.3 Server Menu  
Name  
Possible Values  
Description  
a
Server Type  
When set to FTP, the phone will use the File  
Transfer Protocol (FTP) to obtain configuration  
and phone application files from the boot server.  
FTP, Trivial FTP  
When set to Trivial FTP, the phone will use the  
Trivial File Transfer Protocol (TFTP) to obtain  
configuration and phone application files from  
the boot server.  
Server Address  
dotted-decimal IP address  
OR  
The boot server to use if the DHCP client is dis-  
abled, or the DHCP server does not send a boot  
server option, or the Boot Server parameter is set  
to Static.  
domain name string  
b
any string  
any string  
When the Server Type parameter is set to FTP,  
this is the user name used when the phone logs  
into the FTP server.  
FTP User  
b
When the Server Type parameter is set to FTP,  
this is the password used when the phone logs in  
to the FTP server.  
FTP Password  
a. Using TFTP will make management of the phone more difficult. For more information, see  
2.2.2.1.1 FTP vs. TFTP on page 10.  
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b. The FTP server user name and FTP password should be changed from the default values.  
Note that FTP is an insecure protocol and the user chosen should have very few privileges  
on the server.  
2.2.1.3 Reset to Factory Defaults  
The basic network configuration referred to in the preceding sections can be reset to  
factory defaults. To perform this function on the IP 300, 500 and 600, simultaneously  
press and hold the 4, 6, 8 and * dial pad keys until the password prompt appears. To  
perform this function on the IP 4000, simultaneously press and hold the 6, 8 and * dial  
pad keys until the password prompt appears. Enter the administrator password to ini-  
tiate the reset. This will reset the administrator password as well.  
2.2.2 Application Configuration  
While it is possible to make calls with the phone using its default configuration, most  
installations will require some basic configuration changes to get things running opti-  
mally. These changes can be made using the central boot server model, if a boot  
server has been set up, or some, but not all changes can be made using the phone’s  
internal configuration web server.  
Advantages of using a boot server:  
1. The centralized repository for application images and configuration files permits  
application updates and coordinated configuration parameters.  
2. Some parameters can only be modified using boot server configuration  
files.  
3. The multilingual feature requires boot server-resident dictionary files.  
4. The customized sound effect wave files require a boot server.  
5. When file uploads are permitted, the boot server is the repository for:  
• boot process and application event log files - very effective when diag-  
nosing system problems  
• local configuration changes via the <Ethernet address>-phone.cfg boot  
server configuration overrides file - the phone treats the boot server copy  
as the original when booting  
• per-phone contact directory named <Ethernet address>-directory.cfg  
6. The boot server copy of the application images and configuration files can  
be used to “repair” a damaged phone configuration in the same way that  
system repair disks work for PCs.  
The following sections discuss the available configuration options.  
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2.2.2.1 Centralized Configuration  
The phone application consists of an executable image file (sip.ld) and one or more  
XML-format configuration files. In the centrally provisioned model, these files are  
stored on a boot server and cached in the phone. If the boot server is available at boot  
time, the phone will automatically synchronize its configuration cache with the boot  
server: bootROM image, application executable, and configuration files are all  
upgraded this way. The phone requires that a SNTP server be properly configured for  
this feature to work optimally.  
2.2.2.1.1 FTP vs. TFTP  
Note  
If the phone has bootROM version 2.6.0 or greater, the following restrictions do not apply.  
If using a trivial FTP (TFTP) protocol boot server, synchronization with the boot  
server will be based on file names, not file timestamps, as is the case with FTP. Exe-  
cutable upgrades and changes made to configuration files will not be recognized by  
the phone unless their names are changed.  
Example:  
file.ld ꢀfile01.ld (an upgrade to SIP 1.0.1)  
ipmid.cfg ꢀipmid01.cfg  
The master configuration file, which references the other files and is always down-  
loaded and parsed by the phone, will need to change to reflect these name changes.  
Polycom does not recommend TFTP boot servers for actively managed systems.  
File name management is the responsibility of the System Administrator.  
2.2.2.1.2 Configuration Files  
The phone configuration files consist of master configuration files and application  
configuration files.  
2.2.2.1.2.1 Master Configuration Files  
Central provisioning requires that an XML-format master configuration file be located  
in the home directory on the boot server.  
Per-phone Master Configuration File  
If per-phone customization is required (for all applications that require per-phone cus-  
tomization), the file should be named <Ethernet address>.cfg, where Ethernet address  
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is the Ethernet MAC address of the phone in question. For A-F hexadecimal digits,  
use lower case only, for example, 0004f200106c.cfg. The Ethernet address can be  
viewed using the About soft key during the auto-boot countdown of the bootROM or  
via the Menu>System Status>General menu in the application. It is also printed on a  
label on the back of the phone.  
Default Master Configuration File  
For systems in which the configuration is identical for all phones (no per-phone  
<Ethernet address>.cfg files), the default master configuration file may be used to set  
the configuration for all phones. The file named 000000000000.cfg (<12 zeros>.cfg)  
is the default master configuration file and it is recommended that one be present on  
the boot server. If a phone does not find its own <Ethernet address>.cfg file, it will  
use this one, and establish a baseline configuration. This file is part of the standard  
Polycom distribution of configuration files. It should be used as the template for the  
<Ethernet address>.cfg files.  
The default SIP master configuration file, 000000000000.cfg, is shown below:  
Example:  
<?xm l version= "1.0" standalone="yes" ?>  
<!-- D efault M aster S IP C onfiguration File -->  
<!-- E dit and renam e this file to <E thernet-address>.cfg for each  
phone.-->  
<!-- $R evision: 1.24 $ $D ate: M ar 26 2003 11:59:02 $ -->  
<APPLICATION APP_FILE_PATH="sip.ld"  
CONFIG_FILES="phone1.cfg, sip.cfg, ipm id.cfg"  
M ISC_FILES="" LOG_FILE_DIRECTORY=""/>  
Master configuration files contain four XML attributes:  
APP_FILE_PATH  
CONFIG_FILES  
MISC_FILES  
The path name of the application executable. Has a maximum length  
of 127 characters.  
A comma-separated list of configuration files. Has a maximum length  
of 511 characters, including commas and white space.  
a
A comma-separated list of other required files.  
LOG_FILE_DIRECTORY An alternative directory to use for log files if required. This is left  
blank by default.  
a. MISC_FILES is not used for the SIP application, but is used for the MGCP application.  
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Note  
The order of the configuration files listed in CONFIG_FILES is significant.  
•
•
•
The files are processed in the order listed (left to right).  
The same parameters may be included in more than one file.  
The parameter found first in the list of files will be the one that is effective.  
This provides a convenient means of overriding the behavior of one or more phones without  
altering the baseline configuration files for an entire system.  
2.2.2.1.2.2 Application Configuration Files  
Typically, the files are arranged in the following manner although parameters may be  
moved around within the files and the file names themselves can be changed as  
needed.  
Per-phone settings ꢀ  
Application settings ꢀ  
Core settings ꢀ  
phoneXXXX.cfg  
sip.cfg  
ipmid.cfg  
Category  
Description  
Example  
Core  
Contain parameters that affect the basic operation of the phone  
such as voice codecs, gains, and tones. All phones in an installa-  
tion usually share this category of files.  
ipmid.cfg  
Application-  
specific  
Contain parameters that dictate performance of a particular phone  
application.  
sip.cfg,  
mgcp.cfg  
Typical parameters include the IP address of an application server.  
All phones in an installation usually share application-specific  
files. This file would normally be modified from Polycom tem-  
plates.  
User  
Contain 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.  
12  
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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.  
2.2.2.1.3 Deploying a Boot Server for the Phones  
The following table describes the steps required for successful deployment of a boot  
server for SoundPoint® IP and SoundStation® IP phones. These instructions assume  
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Installation and Operation  
an FTP protocol boot server. For more information on TFTP boot servers, see  
2.2.2.1.1 FTP vs. TFTP on page 10.  
Step:  
Instructions:  
1. Set-up FTP Server:  
Install FTP server application or locate suitable exist-  
ing server. Use RFC-compliant FTP servers. The fol-  
lowing FTP servers have been tested with  
Note: Typically all phones are configured  
with the same FTP account, but the FTP  
account provides a means of conveniently  
partitioning the configuration. Give each  
account an unique home directory on the  
server and change the configuration on an  
account-by-account basis.  
®
®
SoundPoint IP and SoundStation IP and are known  
to work acceptably:  
•
Linux: ProFTPd 1.2.2 through 1.2.9 rc2p, ftpd-  
bsd-0.3.3 (Linux port), we-ftpd 2.6.0  
•
Windows 2000 Server: IIS 5.0, WFTPD 2.03  
a
Create account and home directory.  
2.  
3.  
Copy all files:  
Copy all files from the distribution.zip file to the  
phone home directory. Maintain the same folder hier-  
archy.  
Create per-phone configuration  
Obtain a list of phone Ethernet addresses (barcoded  
label on underside of phone).  
b
files :  
Create per-phone phoneXXXX.cfg and <Ethernet  
address>.cfg files by using the 00000000000.cfg and  
phone1.cfg files from the distribution as templates.  
Edit contents of phoneXXXX.cfg as appropriate. For  
example, edit the registration parameters.  
Edit the CONFIG_FILES attribute of the <Ethernet  
address>.cfg files so that it references the appropriate  
phoneXXXX.cfg file. (Replace the reference to  
phone1.cfg with phoneXXXX.cfg.)  
4.  
5.  
Edit <application>.cfg:  
Edit ipmid.cfg:  
See 4.6.2 SIP Configuration - sip.cfg on page 113,  
particularly for SIP server address.  
See 4.6.1 Core Configuration - ipmid.cfg on page 65.  
Most of the default settings are typically adequate,  
however, if overriding SNTP settings are not available  
via DHCP, the SNTP GMT offset and (possibly) the  
SNTP server address will need to be edited for the 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  
alter its signalling port if local security policy dictates.  
Change the default location settings:  
•
•
user interface language  
time and date format  
14  
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Step:  
Instructions:  
6.  
Decide on FTP server security pol-  
icy:  
Polycom recommends allowing file uploads to the  
FTP server where the security environment permits.  
This allows event log files to be uploaded and changes  
made by the phone user to the configuration (via the  
web server and local user interface) and changes made  
to the directory to be backed up.  
For organizational purposes, configuring a separate  
log file directory is recommended, but not required  
(see LOG_FILE_DIRECTORY in 2.2.2.1.2.1 Master  
Configuration Files on page 10). File permissions  
should give the minimum access required, and the  
account used should have no other rights on the server.  
The phone's FTP account needs to be able to add files  
to which it can write in the log file directory and the  
root directory. It must also be able to list files in all  
directories mentioned in the [mac].cfg file. All other  
files that the phone needs to read, such as the applica-  
tion executable and the standard configuration files,  
should be made read-only via file server file permis-  
sions.  
7.  
Reboot phones after configuring  
their boot server via DHCP or stati-  
cally:  
See 2.2.1 Basic Network Setup on page 4.  
To reboot phones manually, press and hold the follow-  
ing keys simultaneously until a confirmation tone is  
heard or for about three seconds:  
IP 300: Volume-, Volume+, Hold and Redial  
IP 500: Volume-, Volume+, Hold, and Messages  
IP 600: Volume-, Volume+, Mute, and Messages  
IP 4000: *, #, Volume+, and Select  
Monitor the boot server event log and the uploaded  
event log files (if permitted):  
Ensure that the configuration process completed cor-  
rectly.  
Start making calls!  
a. The FTP account name and password must match those configured in the phones them-  
selves. Defaults are: name: PlcmSpIp, password: PlcmSpIp  
b. This step may be omitted if per-phone configuration is not needed.  
2.2.2.2 Local Phone Configuration  
As the only method of modifying phone configuration or as a distributed method of  
augmenting a centralized provisioning model, a local phone-based configuration web  
server is available, unless disabled via ipmid.cfg. For more information, see 4.6.1.9  
Web Server <HTTPD/> on page 91. The phone’s local user interface also permits  
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many application settings to be modified, such as SIP server address or ring type or  
regional settings such as time/date format and language.  
Local Web Server Access  
Point your web browser to http://<phoneIPAddress>/.  
Configuration pages are accessible from the menu along the top 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.  
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:  
Network Configuration  
Authentication  
Requires administrator password.  
Requires user password.  
Reset User Settings  
Requires user password.  
Changes made via the web server or local user interface are stored internally as over-  
rides. These overrides take precedence over settings contained in the configuration  
obtained from the boot server that existed previously within the phone.  
If the boot server permits uploads, these override setting will be saved in a file called  
<Ethernet address>-phone.cfg on the boot server.  
Important  
Local configuration changes will continue to override the boot server-derived configuration  
until deleted via the Reset User Settings menu selection.  
16  
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Administrator 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 65.  
3.1 Basic Features  
3.1.1 Call Log  
The phone maintains a call log. The log:  
• contains call information such as remote party identification, time and date, and  
call duration,  
• allows for convenient redialing of previous outgoing calls and for returning  
incoming calls,  
• can be used to save contact information from call log entries to the contact  
directory.  
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  
ipmid.cfg  
(boot  
•
For more information, see 4.6.1.21 Feature <feature/> on  
page 109.  
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.  
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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. 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  
ipmid.cfg  
(boot  
•
For more information, see 4.6.1.2 User Preferences  
<user_preferences/> on page 69.  
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 via the Reset  
User Settings 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 which are counted as “missed” can be configured  
per registration. Remote missed-call notification can be used to notify the phone when  
a call originally destined for it is diverted by another entity such as a SIP server.  
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Features  
Configuration file:  
ipmid.cfg  
Turn this feature on or off.  
For more information, see 4.6.1.21 Feature <fea-  
ture/> on page 109.  
•
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, see 4.6.3.2.3 Missed Call  
Configuration <serverMissedCall/> on page 123.  
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:  
ipmid.cfg  
Set the key scrolling timeout, key functions, and sub-point-  
ers for each key (usually not necessary).  
Central  
(boot  
server)  
•
For more information, see 4.6.1.13 Keys <keys/> on  
page 96.  
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 500, IP 600 and SoundStation® IP 4000 models.  
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Features  
SoundPoint IP 300 SIP 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 500 SIP 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  
Conference  
Transfer  
Redial  
30  
29  
37  
36  
ABC  
DEF  
Messages  
1
2
3
24  
23  
22  
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  
20  
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Features  
SoundPoint IP 600 SIP Key Layout  
34  
35  
42  
1
33  
41  
2
5
3
4
31  
6
Menu  
Directories  
Services  
Conference  
Transfer  
Redial  
30  
29  
32  
37  
36  
7
8
28  
27  
26  
25  
Messages  
ABC  
DEF  
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  
SoundStation IP 4000 SIP Key Layout  
6
25  
12  
18  
26  
27  
28  
29  
4
1
7
2
8
3
5
10  
9
16  
22  
13  
19  
14  
20  
15  
21  
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Features  
Key ID  
1
IP 300 Function  
Line1  
IP 500 Function  
ArrowUp  
ArrowLeft  
Select  
IP 600 Function  
ArrowUp  
ArrowLeft  
ArrowDown  
ArrowRight  
Select  
IP 4000 Function  
Dialpad1  
Dialpad2  
Dialpad3  
VolUp  
2
Line2  
3
n/a  
4
n/a  
ArrowRight  
ArrowDown  
Delete  
5
Hold  
Handsfree  
ArrowUp  
Dialpad4  
Dialpad5  
Dialpad6  
VolDown  
n/a  
6
n/a  
Delete  
7
DoNotDisturb  
VolUp  
Menu  
Menu  
8
Messages  
DoNotDisturb  
Hold  
Messages  
DoNotDisturb  
MicMute  
VolUp  
9
VolDown  
DialpadStar  
Dialpad0  
DialpadPound  
Dialpad9  
Dialpad8  
Dialpad7  
Dialpad4  
Dialpad5  
Dialpad6  
Dialpad3  
Dialpad2  
Dialpad1  
n/a  
10  
11  
12  
13  
14  
15  
16  
17  
18  
19  
20  
21  
22  
23  
24  
25  
26  
27  
28  
29  
30  
VolUp  
VolDown  
DialpadPound  
Dialpad0  
DialpadStar  
Dialpad9  
Dialpad8  
Dialpad7  
Dialpad4  
Dialpad5  
Dialpad6  
Dialpad3  
Dialpad2  
Dialpad1  
SoftKey4  
SoftKey3  
SoftKey2  
SoftKey1  
Conference  
CallHistory  
VolDown  
DialpadPound  
Dialpad0  
DialpadStar  
Dialpad9  
Dialpad8  
Dialpad7  
Dialpad4  
Dialpad5  
Dialpad6  
Dialpad3  
Dialpad2  
Dialpad1  
SoftKey4  
SoftKey3  
SoftKey2  
SoftKey1  
Services  
Select  
Dialpad7  
Dialpad8  
Dialpad9  
MicMute  
n/a  
ArrowDown  
DialpadStar  
Dialpad0  
DialpadPound  
Redial  
Redial  
n/a  
n/a  
n/a  
SoftKey3  
MicMute  
SoftKey2  
SoftKey1  
ArrowDown  
n/a  
Menu  
Exit  
SoftKey1  
SoftKey2  
SoftKey3  
n/a  
Directories  
22  
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Features  
Key ID  
31  
IP 300 Function  
IP 500 Function  
Services  
Directories  
Line3  
IP 600 Function  
Line6  
IP 4000 Function  
ArrowUp  
Menu  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
n/a  
32  
Conference  
Line2  
33  
34  
n/a  
Line2  
Line1  
35  
Headset  
n/a  
Line1  
Line3  
36  
Redial  
Redial  
37  
n/a  
Transfer  
Headset  
MicMute  
Handsfree  
n/a  
Transfer  
Headset  
Handsfree  
Hold  
38  
n/a  
39  
n/a  
40  
n/a  
41  
n/a  
Line4  
42  
n/a  
n/a  
Line5  
3.1.8 Connected Party Identification  
Where possible, the identity of the remote party to which the user has connected is dis-  
played and logged . The connected party identity is derived from the network signal-  
ing. In some cases the remote party will be different from the called party identity due  
to network call diversion.  
3.1.9 Context Sensitive Volume Control  
The volume of user interface sound effects, such as the ringer, and the receive volume  
of call audio is adjustable. While transmit levels are fixed according to the TIA/EIA-  
810-A standard, receive volume is adjustable. For SoundPoint® IP, if using the default  
configuration parameters, the receive handset/headset volume resets to nominal after  
each call to comply with regulatory requirements. See 4.6.1.6.2 Volume Persistence  
<volume/> on page 80.  
3.1.10 Customizable Audio Sound Effects  
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  
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Features  
audio files. The default sample audio files may be replaced with alternates in .wav file  
format. Supported .wav formats include:  
• mono G.711 (13-bit dynamic range, 8-khz sample rate),  
• mono L16/1600 (16-bit dynamic range, 16-kHz sample rate).  
Note  
The alternate sampled audio sound effect files must be present on the boot server or the Internet for  
downloading at boot time.  
Configuration File: Specify patterns used for sound effects and the individual  
ipmid.cfg  
tones or sampled audio files used within them.  
For more information, see:  
Central  
(boot  
server)  
•
•
•
4.6.1.1.3 Call Progress Tones <callProgTones> on  
page 68  
4.6.1.4 Sampled Audio for Sound Effects  
<sampled_audio/> on page 72  
4.6.1.5 Sound Effects <sound_effects/> on page 73  
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 via the Reset User Set-  
tings menu selection.  
Local  
Local Phone User  
Interface  
None.  
3.1.11 Message Waiting Indication  
The phone will flash a message-waiting indicator LED when instant messages are  
waiting, and it can be configured to do so when voice messages are waiting.  
3.1.12 Distinctive Incoming Call Treatment  
The phone can automatically apply distinctive treatment to calls containing specific  
attributes. The distinctive treatment that can be applied includes customizable alerting  
sound effects and automatic call diversion or rejection. Call attributes that can trigger  
distinctive treatment include the calling party name or SIP contact (number or URL  
format).  
24  
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Administration: Distinctive Incoming Call Treatment  
For more information, see 3.1.16 Local Contact Directory on page 27.  
3.1.13 Distinctive Ringing  
There are three aspects to Distinctive Ringing:  
1. The user can select the ring type for each line. There are many different ring pat-  
terns to choose from.  
2. The ring type for specific callers can be assigned in the contact directory.  
For more information, see 3.1.12 Distinctive Incoming Call Treatment on  
page 24. This feature has higher priority than Item 1.  
3. The SIP Alert-Info field can be used to map calls to specific ring types.  
This feature has higher priority than Items 1 and 2.  
Configuration file:  
sip.cfg  
Specify the mapping of Alert-Info strings to ring types.  
•
For more information, see 4.6.2.1.3.2 Alert Infor-  
mation <alertInfo/> on page 116.  
Central  
(boot  
server)  
XML File: <Ethernet  
address>-direc-  
tory.xml  
This file can be created manually using an XML editor.  
•
For more information, see 3.1.16.1 Local Contact  
Directory File Format on page 28.  
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 via the Reset User Settings menu selection.  
3.1.13 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, see 4.6.2.1.3.2 Alert Informa-  
tion <alertInfo/> on page 116.  
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Features  
Web Server  
None.  
None.  
(if enabled)  
Local  
Local Phone User  
Interface  
3.1.14 Do-Not-Disturb  
A do-not-disturb feature is available to temporarily stop all incoming call alerting.  
Calls can optionally be treated as though the phone is busy while Do-Not-Disturb  
(DND) is enabled. Incoming calls received while DND is enabled are logged as  
missed.  
Configuration file:  
ipmid.cfg  
Specify whether or not DND results in incoming calls  
being given busy treatment.  
•
For more information, see 4.6.1.10 Call Handling  
Configuration <call/> on page 92.  
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, see 4.6.3.2.1 Do Not Disturb  
<donotdisturb/> on page 123.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
Enable or disable DND using the “Do Not Disturb” key  
on the IP 300, 500 and 600 or the Features menu on the  
IP 4000.  
3.1.15 Handset, Headset, and Speakerphone  
SoundPoint® IP phones come standard with a handset and a dedicated connector is  
provided for a headset (not supplied). The IP 500 and IP 600 phones have full-duplex  
speakerphones. The IP 300 has a listen-only speakerphone. The SoundPoint® phones  
provide dedicated keys for convenient selection of either the speakerphone or headset.  
The SoundStation® IP 4000 phones are full-duplex speakerphones.  
Configuration file:  
ipmid.cfg  
Enable or disable persistent headset mode.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.2 User Preferences  
<user_preferences/> on page 69.  
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Web Server  
(if enabled)  
None.  
Local Phone User  
Interface  
Enable or disable persistent headset mode via the Settings  
menu. Changes are saved to local flash and backed up to  
<Ethernet address>-phone.cfg on the boot server.  
Changes will permanently override global settings unless  
deleted via the Reset User Settings menu.  
Local  
3.1.16 Local Contact Directory  
The phone maintains a local contact directory. The directory can be downloaded from  
the boot server and edited locally. Contact information from previous calls may be  
easily added to the directory for convenient future access. The directory is the central  
database for several other features including speed-dial, distinctive incoming call  
treatment, presence, and instant messaging.  
Configuration file:  
ipmid.cfg  
Set whether the directory uses volatile storage on the  
phone (required on the IP 500 platform for directories  
greater than 25 entries).  
•
For more information, see 4.6.1.11 Directory  
<directory/> on page 93.  
XML file:  
000000000000-direc-  
tory.xml  
A sample file named 000000000000-directory~.xml  
(Note extra “~” in the file name) is included with the  
application file distribution. This file can be used as a  
template for the per-phone <Ethernet address>-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 than 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.  
Central  
(boot  
server)  
•
For information on file format, see 3.1.16.1 Local  
Contact Directory File Format on page 28.  
XML file: <Ethernet  
This file can be created manually using an XML editor.  
address>-directory.xml  
•
For information on file format, see 3.1.16.1 Local  
Contact Directory File Format on page 28.  
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Features  
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  
3.1.16.1 Local Contact Directory File Format  
An example local contact directory is shown. Look to the table for an explanation of  
each element.  
Local Contact Directory File example:  
<?xml version="1.0" encoding="UTF-8" standalone="yes" ?>  
<directory>  
<item_list>  
<item>  
<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>  
28  
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Features  
Element  
Permitted Values  
Interpretation  
fn  
UTF-8 encoded string of up to  
first name  
a
40 bytes  
ln  
ct  
UTF-8 encoded string of up to  
40 bytes  
last name  
contact  
UTF-8 encoded string contain-  
ing digits (the user part of a SIP  
URL) or a string that constitutes  
a valid SIP URL  
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 40  
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.  
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3.1.17 Local Digit Map  
The phone has a local digit map feature to automate the setup phase of number-only  
calls. When properly configured, this feature eliminates the need for using the Send  
soft key when making outgoing calls. Instead, as soon as a digit pattern matching the  
digit map is found, the call setup process will complete automatically. This feature is  
similar to the digit map feature of the Media Gateway Control Protocol (MGCP) and  
the 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, see 4.6.2.1.3.5 Conference  
Setup <conference/> on page 118.  
Central  
(boot  
server)  
Configuration file:  
phone1.cfg  
Specify per-registration impossible match behavior, trail-  
ing # behavior, digit map matching strings, and time out  
values that override those in sip.cfg.  
•
For more information, see 4.6.3.4 Dial Plan <dial-  
plan/> on page 126.  
Web Server  
(if enabled)  
Specify 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 via the Reset User Settings menu selection.  
Local  
Local Phone User  
Interface  
None.  
3.1.18 Microphone Mute  
A microphone mute feature is provided. When activated, visual feedback is provided.  
This is a local function and cannot be overridden by the network.  
3.1.19 Multiple Call Appearances  
The phone supports multiple concurrent calls. The hold feature can be used to pause  
activity on one call and switch to another call. When active on one call, an additional  
incoming call is presented using the familiar “call waiting” style. Soft keys with call  
disposition options are presented to the user. The current user interface is limited to  
30  
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two concurrent calls per registration (line). This is an artificial limit and will be  
expanded in the future.  
3.1.20 Shared Call Appearances  
Calls and lines on multiple phones can be logically related to each other. A call that is  
active on one phone will be presented visually to phones which share that call 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 which binds the appearances  
together logically and looks after the necessary state notifications and performs an  
access control function. For more information, see 5.2.4 Shared Call Appearance Sig-  
naling on page 139.  
Configuration file:  
ipmid.cfg  
Specify whether diversion should be disabled on shared  
lines.  
•
For more information, see 4.6.1.10 Call Handling  
Configuration <call/> on page 92.  
Configuration file:  
sip.cfg  
Specify line-seize subscription period.  
•
For more information, see 4.6.2.1.2 Server <server/>  
on page 114.  
Specify standard or non-standard behavior for processing  
line-seize subscription for mutual exclusion feature.  
Central  
(boot  
server)  
•
For more information, see 4.6.2.1.3.4 Special Events  
<specialEvent/> on page 118.  
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, see 4.6.3.1 Registration <reg/>  
on page 121.  
Specify per-registration whether diversion should be dis-  
abled on shared lines.  
•
For more information, see 4.6.3.3 Diversion <divert/>  
on page 124.  
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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 via the  
Reset User Settings 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.  
3.1.21 Bridged Line Appearances  
Calls and lines on multiple phones can be logically related to each other. A call that is  
active on one phone will be presented visually to phones which share that line. Mutual  
exclusion features emulate traditional PBX or key system privacy for shared calls.  
Incoming calls can be presented to multiple phones simultaneously. This feature is  
dependent on support from a SIP server which binds the appearances together logi-  
cally and looks after the necessary state notifications and performs an access control  
32  
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function. For more information, see 5.2.5 Bridged Line Appearance Signaling on  
page 139.  
Note: In the configuration files, bridged lines are configured by “shared line” parame-  
ters.  
Configuration file:  
ipmid.cfg  
Specify whether diversion should be disabled on shared  
lines.  
•
For more information, see 4.6.1.10 Call Handling  
Configuration <call/> on page 92.  
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, see 4.6.3.1 Registration <reg/>  
on page 121.  
Specify per-registration whether diversion should be dis-  
abled on shared lines.  
•
For more information, see 4.6.3.3 Diversion <divert/>  
on page 124.  
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 via the  
Reset User Settings 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.  
3.1.22 Customizable Fonts and Indicators  
The phone’s user interface can be customized by changing the fonts and graphic icons  
used on the display and the LED indicator patterns. Pre-existing fonts embedded in  
the software can be overwritten or new fonts can be downloaded. The bitmaps and bit-  
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Features  
map animations used for graphic icons on the display can be changed and reposi-  
tioned. LED flashing sequences and colors can be changed.  
Configuration File: Specify fonts to overwrite existing ones or specify new fonts.  
ipmid.cfg  
•
For more information, see 4.6.1.12 Fonts <font/> on  
page 94.  
Specify which bitmaps to use.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.14 Bitmaps <bitmaps/>  
on page 100.  
Specify how to create animations and LED indicator patterns.  
•
For more information, see 4.6.1.15 Indicators <indicators/  
> on page 100.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
3.1.23 Soft Key-Driven User Interface  
The user interface makes extensive use of intuitive, context-sensitive soft key menus.  
3.1.24 Speed Dial  
Entries in the local directory can be linked to the speed dial system. The speed dial  
system allows calls to be placed quickly from dedicated keys as well as from a speed  
dial menu.  
XML file:  
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, see 3.1.16.1 Local  
Contact Directory File Format on  
page 28.  
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Features  
Web Server (if enabled)  
None.  
Local Phone User Interface  
The user can edit the directory contents. The  
Speed Dial Index field is used to link direc-  
tory 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.  
3.1.25 Time and Date Display  
The phone maintains a local clock and calendar. Time and date can be displayed in  
certain operating modes such as when the phone is idle. The clock and calendar must  
be synchronized to a remote SNTP timeserver.  
Configuration file:  
ipmid.cfg  
Turn time and date display on or off.  
•
For more information, see 4.6.1.2 User Preferences  
<user_preferences/> on page 69.  
Set the time and date display formats.  
•
For more information, see 4.6.1.1.2 Date and Time  
<datetime/> on page 68.  
Central  
(boot  
server)  
Set the basic SNTP settings.  
•
For more information, see 4.6.1.8.2 Time Synchro-  
nization <SNTP/> on page 88.  
Set daylight savings parameters.  
•
For more information, see 4.6.1.8.2 Time Synchro-  
nization <SNTP/> on page 88.  
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Features  
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  
via the Reset User Settings menu selection.  
Local Phone User  
Interface  
The basic SNTP settings can be made in the Network  
Configuration menu.  
Local  
•
For more information, see 2.2.1.1 DHCP or Manual  
TCP/IP Setup on page 5.  
The user can edit the time and date format under the Set-  
tings 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  
via the Reset User Settings menu selection.  
3.1.26 Idle Display Animation  
All phones except the SoundPoint® IP 300 can display a customized animation on the  
idle display in addition to the time and date. For example, a company logo could be  
displayed.  
Configuration file:  
ipmid.cfg  
To turn idle display animation on or off.  
•
For more information, see 4.6.1.15 Indicators  
<indicators/> on page 100.  
To replace the animation used for the idle display.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.15.1 Animations  
<Animations/> <IP_300/>, <IP_500/>, <IP_600/>  
and <IP_4000/> on page 101.  
To change the position of the idle display animation.  
•
For more information, see 4.6.1.15.4.2 Graphic  
Icons <gi/> <IP_300/>, <IP_500/>, <IP_600/> and  
<IP_4000/> on page 102.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
36  
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Features  
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, see 4.6.3.2.2 Automatic Off-  
hook Call Placement <autoOffHook/> on page 123.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
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 signalling is employed to request that the remote party  
stop sending media and to inform them that they are being held. A configurable local  
hold reminder feature can be used to remind the user that they have placed calls on  
hold.  
Configuration file: Specify whether RFC 2543 (c=0.0.0.0) or RFC 3264 (a=sen-  
sip.cfg  
donly or a=inactive) outgoing hold signalling is used.  
•
For more information, see 4.6.2.1.3 SIP <SIP/> on  
page 115.  
Central  
(boot  
server)  
Configuration file: Specify local hold reminder options.  
ipmid.cfg  
•
For more information, see 4.6.1.10.1 Hold, Local  
Reminder <localReminder/> on page 93.  
Web Server  
(if enabled)  
Specify whether or not to use RFC 2543 (c=0.0.0.0) outgo-  
ing hold signalling. The alternative is RFC 3264 (a=sen-  
donly or a=inactive). .  
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 signalling. The  
alternative is RFC 3264 (a=sendonly or a=inactive).  
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Features  
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 both blind transfers (call is trans-  
ferred to User C without first consulting privately with User C) and transfers with  
consultation (call is transferred to User C after first consulting privately with User C).  
3.2.4 Three-Way Conference, Local or Centralized  
1
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 depends 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, see 4.6.2.1.3.5 Conference Setup  
<conference/> on page 118.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
3.2.5 Call Diversion (Call Forward)  
The phone provides a flexible call diversion feature to divert (forward) calls to another  
destination. Call diversion can be applied automatically to all calls, only when the  
phone is busy, or after an extended period of alerting. The user can elect to manually  
divert calls while they are in the alerting state to a predefined or manually specified  
destination. The call diversion feature works in conjunction with the distinctive  
®
1. On SoundStation IP 4000, conferences are not available if the G.729 codec is enabled on the phone.  
This restriction will be removed in future releases.  
38  
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Features  
incoming call treatment feature. The user’s ability to originate calls is unaffected by  
all call diversion options. Each registration (line) has its own diversion properties.  
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, see 4.6.3.3 Diversion <divert/>  
on page 124.  
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 via the Reset  
User Settings 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 via the Reset  
User Settings 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:  
ipmid.cfg  
Turn this feature on or off.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.21 Feature <feature/>  
on page 109.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
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Features  
3.2.7 Group Call Pick-up  
Calls to another phone within a pre-defined group can be picked up without dialing the  
extension of the other phone. This feature depends on support from a SIP server.  
Configuration file:  
ipmid.cfg  
Turn this feature on or off.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.21 Feature <feature/>  
on page 109.  
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:  
ipmid.cfg  
Turn this feature on or off.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.21 Feature <feature/>  
on page 109.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
40  
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Features  
3.2.9 Last Call Return  
The phone allows server-based last call return. This feature depends on support from a  
SIP server.  
Configuration file:  
ipmid.cfg  
Turn this feature on or off.  
•
For more information, see 4.6.1.21 Feature <feature/>  
on page 109.  
Central  
(boot  
server)  
Specify the string sent to the server for last-call-return.  
•
For more information, see 4.6.1.10 Call Handling  
Configuration <call/> on page 92.  
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-  
urable for different network environments. When packets are lost, a concealment  
algorithm minimizes the resulting negative audio consequences.  
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Configuration file: Set the jitter buffer tuning parameters including minimum  
Central  
(boot  
server)  
ipmid.cfg  
and maximum size and shrink aggression.  
For more information, see 4.6.1.6.1.2 Codec Profiles  
<profiles/> on page 79.  
•
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 per-  
manently override global settings unless deleted via the  
Reset User Settings menu selection.  
Local  
Local Phone User  
Interface  
None.  
3.3.3 Local Conference Mixing  
The phone’s audio processing subsystem contains a flexible three-party conferencing  
2
capability . This feature can be used to set up local three-party conferences where no  
external protocol signaling is involved.  
3.3.4 Voice Activity Detection (VAD)  
The purpose of VAD is to conserve network bandwidth by detecting periods of rela-  
tive “silence” in the transmit data path and replacing that silence efficiently with spe-  
cial 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 com-  
fort noise (CN) payload format (or bit-stream) for G.711 use in packet-based, multi-  
media communication systems. The phone generates CN packets (also known as  
Silence Insertion Descriptor (SID) frames) and also decodes CN packets, efficiently  
regenerating a facsimile of the background noise at the remote end.  
Configurationfile: Enable or disable VAD and set the detection threshold.  
Central  
ipmid.cfg  
(boot  
•
For more information, see 4.6.1.6.10 Voice Activity  
Detection <VAD/> on page 86.  
server)  
®
42  
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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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Features  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
3.3.5 DTMF Tone Generation  
The phone generates DTMF tones in response to user dialing on the dial pad. These  
tones are transmitted in the RTP streams of connected calls. The phone can encode the  
DTMF tones using the active voice codec or using RFC 2833 compatible encoding.  
The coding format decision is based on the capabilities of the remote endpoint.  
Configuration file: Set the DTMF tone levels, autodialing on and off times, and  
Central  
(boot  
server)  
ipmid.cfg  
other parameters.  
For more information, see 4.6.1.3.1 Dual Tone Multi-  
Frequency <DTMF/> on page 70.  
•
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
3.3.6 DTMF Event RTP Payload  
The phone is compatible with RFC 2833 - RTP Payload for DTMF Digits, Telephony  
Tones, and Telephony Signals. RFC 2833 describes a standard RTP-compatible tech-  
nique for conveying DTMF dialing and other telephony events over an RTP media  
stream. The phone generates RFC 2833 (DTMF only) events but does not regenerate,  
nor otherwise use, DTMF events received from the remote end of the call.  
Configuration file: Enable or disable RFC 2833 support in SDP offers and spec-  
Central  
(boot  
server)  
ipmid.cfg  
ify the payload value to use in SDP offers.  
•
For more information, see 4.6.1.3.1 Dual Tone Multi-  
Frequency <DTMF/> on page 70.  
Web Server  
(if enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
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3.3.7 Acoustic Echo Cancellation (AEC)  
The phone employs advanced acoustic echo cancellation for hands-free operation.  
Both linear and non-linear techniques are employed to aggressively reduce echo yet  
provide for natural full-duplex communication patterns.  
3.3.8 Audio Codecs  
The following table summarizes the phone’s audio codec support:  
Effective  
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:  
ipmid.cfg  
Specify codec priority, preferred payload sizes, and jitter  
buffer tuning parameters.  
Central  
(boot  
server)  
For more information, see  
•
4.6.1.6.1.1 Codec Preferences <preferences/> on  
page 79, and  
•
4.6.1.6.1.2 Codec Profiles <profiles/> on page 79.  
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 via the  
Reset User Settings menu selection.  
Local  
Local Phone User  
Interface  
None.  
44  
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3.3.9 Background Noise Suppression (BNS)  
This feature, designed primarily for handsfree operation, reduces background noise to  
enhance communication in noisy environments.  
3.3.10 Comfort Noise Fill  
Comfort noise fill is designed to help provide a consistent noise level to the remote  
user of a handsfree call. Fluctuations in perceived background noise levels are an  
undesirable side effect of the non-linear component of most AEC systems. This 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.  
3.3.11 Automatic Gain Control (AGC)  
This feature, applicable to handsfree operation, is used to boost the transmit gain of the  
3
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 and MSN® Messenger version 4.7  
and Windows® Messenger 5.0. The phone’s presence and instant messaging features  
are integrated with the contact directory features, using its contact database.  
3.4.1 Presence  
The Presence feature allows the phone to monitor the status of other users/devices and  
allows other users to monitor it. The status of monitored users is displayed visually  
and is updated in real time. The user can block others from monitoring her phone and  
4
is notified when a change in monitored status occurs . Phone status changes are  
broadcast automatically to monitoring phones when the user engages in calls or  
3. AGC support will be available in a subsequent release.  
4. Notification when a change in monitored status occurs will be available in a subsequent release.  
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Features  
invokes do-not-disturb. The user can also manually specify a state to convey, overrid-  
ing, and perhaps 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, see 3.1.16.1 Local Contact  
Directory File Format on page 28.  
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 have multilingual user interfaces. The System  
Administrator or the user can choose the language. Support for major western Euro-  
pean languages is included and additional languages can be easily added. Support for  
46  
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Asian languages (Chinese, Japanese, and Korean) is also included but will render only  
on the SoundPoint® IP 600’s and SoundStation® IP 4000’s higher resolution displays.  
Basic character support includes the following Unicode character ranges:  
Name  
Range  
C0 Controls and Basic Latin  
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  
5
4000 platforms includes the following Unicode character ranges .  
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 dic-  
tionary files are inaccessible, the language will revert to the internal language.  
5. Within a Unicode range, some characters may not be supported due to their infrequent usage.  
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Note  
Currently, the multilingual feature is only available in the application. At this time, the  
bootROM application is English only.  
Configuration file:  
ipmid.cfg  
Specify the boot-up language and the selection of language  
choices to be made available to the user.  
Central  
(boot  
server)  
For more information, see:  
•
•
4.6.1.1.1 Multilingual <multilingual/> on page 66, and  
4.6.1.1.1.1 Adding New Languages on page 67.  
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 via the Reset User Settings menu selection.  
Local  
3.5.2 Downloadable Fonts  
New fonts can be loaded onto the phone. For more information, see 4.6.1.12 Fonts  
<font/> on page 94.  
3.5.3 Synthesized Call Progress Tones  
In order to emulate the familiar and efficient audible call progress feedback generated  
by the PSTN and traditional PBX equipment, call progress tones are synthesized dur-  
48  
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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:  
ipmid.cfg  
Specify the basic tone frequencies, levels, and basic  
repetitive cadences.  
•
For more information, see 4.6.1.3.2 Chord Sets  
<chord_sets/> on page 71 and 4.6.1.1.3 Call  
Progress Tones <callProgTones> on page 68.  
Specify downloaded sampled audio files for advanced  
call progress tones.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.4 Sampled Audio  
for Sound Effects <sampled_audio/> on page 72.  
Specify patterns.  
For more information, see:  
•
•
4.6.1.5.1 Patterns <patterns/> on page 74, and  
4.6.1.5.1.1 Call Progress Patterns on page 75.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
3.6 Advanced Server Features  
3.6.1 Voicemail Integration  
The phone is compatible with voicemail servers. The subscribe contact and callback  
mode can be configured per user/registration on the phone. The phone can be config-  
ured with a SIP URL to be called automatically by the phone when the user elects to  
retrieve messages. Voicemail access can be configured to be via a single key press if  
only one registration has voicemail set up and the phone has a dedicated function key  
for this purpose (for example the Messages key on the IP 500 and IP 600). A message-  
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waiting signal from a voicemail server will trigger the message-waiting indicator to  
flash.  
Configuration file:  
ipmid.cfg  
For one-touch voicemail access, enable the “one-touch  
voicemail” user preference.  
•
For more information, see 4.6.1.2 User Preferences  
<user_preferences/> on page 69.  
Configuration file:  
phone1.cfg  
For one-touch voicemail access, choose to bypass instant  
messages to remove the step of selecting between instant  
messages and voicemail after pressing the Messages key on  
the IP 500 and IP 600 (instant messages are still accessible  
from the Main Menu).  
Central  
(boot  
server)  
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 voicemail.  
•
For more information, see 4.6.3.5 Messaging <msg/>  
on page 128.  
Web Server  
(if enabled)  
For one-touch voicemail access, enable the “one-touch  
voicemail” user preference and choose to bypass instant  
messages to remove the step of selecting between instant  
messages and voicemail after pressing the Messages key on  
the IP 500 and IP 600 (instant messages are still accessible  
from the Main Menu).  
Navigate to: http://<phoneIPAddress>/coreConf.htm  
On a per-registration basis, specify a subscribe contact for  
solicited NOTIFY applications, a callback mode (self call-  
back or another contact) to call when the user accesses  
voicemail.  
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 via the  
Reset User Settings menu selection.  
Local Phone User  
Interface  
None.  
3.6.2 Multiple Registrations  
SoundPoint® IP phones support multiple registrations per phone and the SoundSta-  
tion® IP 4000 supports a single registration. The SoundPoint® IP 300 supports two  
registrations, the IP 500 supports three and the IP 600 supports six. Each registration  
is mapped to the familiar concept of a phone line. The user can select which line to  
50  
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use for outgoing calls or which registration 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, see 4.6.2.1.1 Local <local/> on  
page 113 and 4.6.2.1.2 Server <server/> on page 114.  
Configuration file: For up to six registrations, specify a display name, a SIP  
Central  
(boot  
server)  
phone1.cfg  
address, an optional display label, an authentication user ID  
and password, and an optional array of registration servers.  
The authentication user ID and password are optional and  
for security reasons can be omitted from the configuration  
files. The local flash parameters will be used instead. The  
optional array of servers and their associated parameters will  
override the servers specified in sip.cfg if non-Null.  
•
For more information, see 4.6.3.1 Registration <reg/>  
on page 121.  
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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  
For up to six registrations, specify a display name, a SIP  
address, an optional display label, an authentication user ID  
and password, and an optional array of registration servers.  
The authentication user ID and password are optional and  
for security reasons can be omitted from the configuration  
files. The local flash parameters will be used instead. The  
optional array of servers will override the servers specified  
in sip.cfg in non-Null. This will also override the servers on  
the appConf.htm web page.  
Navigate to: http://<phoneIPAddress>/reg.htm  
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 via the Reset  
User Settings menu selection.  
Local  
Local Phone User  
Interface  
Use the SIP Configuration menu to specify the local SIP sig-  
naling port, an array of SIP servers to register to and regis-  
tration information for up to six registrations. The SIP  
Configuration menu contains a sub-set of all the parameters  
available in the configuration files.  
Either the Web Server or the boot server configuration files  
or the local phone user interface should be used to configure  
registrations, not a mixture of these options. When the SIP  
Configuration menu is used, it is assumed that all 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 per-  
manently override global settings unless deleted via the  
Reset User Settings menu selection.  
•
For more information on the fields in this menu, see  
4.6.2.1.1 Local <local/> on page 113, 4.6.2.1.2 Server  
<server/> on page 114 and 4.6.3.1 Registration <reg/>  
on page 121.  
3.6.3 ACD login / logout  
The phone allows ACD (Automatic Call Distribution) login and logout. This feature  
depends on support from a SIP server.  
Configuration file:  
ipmid.cfg  
Turn this feature on or off.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.21 Feature <feature/>  
on page 109.  
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Web Server  
None.  
None.  
(if enabled)  
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:  
ipmid.cfg  
Turn this feature on or off.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.21 Feature <feature/>  
on page 109.  
Web Server  
(if enabled)  
None.  
Local  
Local Phone User  
Interface  
None.  
3.6.5 Server Redundancy  
The phone can be configured with multiple SIP servers, one primary and one or more  
backup. The phone will switch to a backup server when the current primary server  
fails. Backup server configuration can be static or can use advanced DNS methods. In  
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 via all servers in the list and this is the last server then  
the signaling fails after the complete UDP timeout defined in RFC 3261. If it is not the  
last server in the list, the maximum number of retries using the configurable retry tim-  
eout is used. For more information, see 4.6.2.1.2 Server <server/> on page 114 and  
4.6.3.1 Registration <reg/> on page 121.  
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3.6.5.1 DNS SIP Server Name Resolution  
If a DNS name is given for a proxy/registrar address, the IP address(es) associated  
with that name will be discovered as specified in RFC 3263 - Locating SIP Servers. If  
a port is given, the only lookup will be an A record. If no port is given, NAPTR and  
SRV records will be tried, before falling back on A records if NAPTR and SRV  
records return no results. If no port is given, and none is found through DNS, 5060  
will be used.  
See http://www.ietf.org/rfc/rfc3263.txt for an example.  
3.7 Accessory Internet Features  
3.7.1 MicroBrowser  
The SoundPoint® IP 600 phone supports an XHTML microbrowser. This can be  
launched by pressing the Services key.  
Configuration file: Specify the Services browser home page, a proxy to use, and  
Central  
(boot  
server)  
ipmid.cfg  
size limits.  
For more information, see 4.6.1.23 MicroBrowser  
<microbrowser/> on page 111.  
•
Web Server  
(if enabled)  
None  
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  
54  
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user password is requested, the administrator password will also work. The web  
server is protected by the administrator password.  
Configuration file:  
ipmid.cfg  
Specify the minimum lengths for the user and administrator  
passwords.  
Central  
(boot  
server)  
•
For more information, see 4.6.1.17.1 Password Lengths  
<pwd/><length> on page 106.  
Web Server  
(if enabled)  
None.  
Local Phone User  
Interface  
The user and administrator passwords can be changed under  
the Settings menu. 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 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, see 4.6.2.1.3.3 Request Valida-  
tion <requestValidation/> on page 117.  
Web Server (if  
enabled)  
None.  
None.  
Local  
Local Phone User  
Interface  
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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 via a local AC power adapter or can be line-  
powered (power supplied via the signaling or idle pairs of the LAN Ethernet cable).  
Line powering typically requires that the phone plugs directly into a dedicated LAN  
jack. Devices that do not require LAN power can then plug into the SoundPoint® 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 RTP  
packet types are set to 2 or greater. Otherwise, these packets will compete equally  
with those from the PC port. For more information, see 4.6.1.7 Quality of Service  
<QOS/> on page 86.  
4.2 Application Network Setup  
4.2.1 RTP Ports  
The phone is compatible with RFC 1889 - RTP: A Transport Protocol for Real-Time  
Applications - and the updated RFCs 3550 and 3551. Consistent with RFC 1889, the  
phone treats all RTP streams as bi-directional from a control perspective and expects  
that both RTP endpoints will negotiate the respective destination IP addresses and  
ports. This allows RTCP to operate correctly even with RTP media flowing in only a  
single direction, or not at all. It also allows greater security: packets from unautho-  
rized 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 mul-  
tiple RTP streams are supported, several ports can be used concurrently. Consistent  
with RFC 1889, the next higher odd port is used to send and receive RTCP.  
Configuration file:  
ipmid.cfg  
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, see 4.6.1.8.3.1 RTP <RTP/> on  
page 90.  
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>/coreConf.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 via the Reset  
User Settings menu selection.  
Local Phone User  
Interface  
None.  
4.2.2 Working with Network Address Translation  
(NAT)  
The phone can work with certain types of network address translation (NAT). The  
phone’s signaling and RTP traffic use symmetric ports (the source port in transmitted  
packets is the same as the associated listening port used to receive packets) and the  
external IP address and ports used by the NAT on the phone’s behalf can be 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, see 4.6.3.6 Network Address  
Translation <nat/> on page 129.  
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>/coreConf.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 via the Reset  
User Settings 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 via the centralized provisioning (boot server) model.  
To automatically update:  
1. Back-up old application and configuration files. The old configuration can be easily  
restored by reverting to the back-up files.  
2. Customize new configuration files or apply new or changed parameters to  
the old configuration files. Differences between old and new versions of  
configuration files are explained in the Release Notes which accompany  
the software.Changes to site-wide configuration files such as ipmid.cfg  
can be done manually, but a scripting tool is useful to change per-phone  
configuration files.  
3. Save the new configuration files and images (such as sip.ld) on the boot  
server. 6  
4. Reboot the phones (see below).  
6. For TFTP, refer to 2.2.2.1.1 FTP vs. TFTP on page 10  
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For more information, see 2.2.2 Application Configuration on page 9.  
Manual Reboot: Menu Option or Key Presses  
To reboot phones manually, a menu option can be selected or a key combination can  
be held down. The menu option is called Restart Phone and it is in the Settings menu.  
For the key combination, press and hold the following keys simultaneously until a  
confirmation tone is heard or for about three seconds:  
IP 300:  
IP 500:  
IP 600:  
IP 4000:  
Volume-, Volume+, Hold, Do Not Disturb  
Volume-, Volume+, Hold, Messages  
Volume-, Volume+, Mute, Messages  
*, #, Volume+, Select  
Centralized Reboot:  
The phones can be rebooted remotely via the SIP signaling protocol. Refer to  
4.6.2.1.3.4 Special Events <specialEvent/> on page 118.  
Periodic Polling For Upgrades:  
The phones can be configured to periodically poll the boot server to check for changed  
configuration files or application executable. If a change is detected the phone will  
reboot to download the change. Refer to 4.6.1.18 Provisioning <provisioning/> on  
page 107.  
4.4 Event Logging  
The phones maintain both boot and application event log files. These files can be  
helpful when diagnosing problems. The event log files are stored in the phone’s flash  
file system and are periodically uploaded to the provisioning boot server if permitted  
by security policy. The files are stored in the phone’s home directory or a user-config-  
urable directory on the boot server. Both overwrite and append modes are supported  
for the application event log.  
The event log files are:  
• <Ethernet address>-boot.log  
• <Ethernet address>-app.log  
The boot log file is uploaded to the boot server after every reboot.  
The application log file is uploaded periodically or when the local copy reaches a pre-  
determined size.  
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As an additional diagnostic tool, both log files can be uploaded on demand to the boot  
server by pressing and holding the following keys until a confirmation tone is heard or  
for about three seconds.  
IP 300:  
IP 500:  
IP 600:  
IP 4000:  
Line1, Line2, Arrow Up, Arrow Down  
The four arrow keys  
The four arrow keys  
Menu, Exit, Off-hook/Handsfree, Redial  
Log files uploaded in this manner are named:  
• <Ethernet address>-now-boot.log  
• <Ethernet address>-now-app.log  
Configuration file:  
ipmid.cfg  
Specify a multitude of event logging settings.  
•
For more information, see 4.6.1.16 Event Logging  
<logging/> on page 103.  
Configuration file:  
sip.cfg  
Specify “sip” event logging settings.  
Central  
(boot  
server)  
•
For more information, see 4.6.2.1.5 SIP Logging  
<logging/> <level/> <change/> on page 120.  
Configuration file:  
<Ethernet  
address>.cfg  
Specify different directory to use for log files if desired.  
•
For more information, see 2.2.2.1.2.1 Master Configu-  
ration Files on page 10.  
Web Server  
(if enabled)  
Specify a multitude of event logging settings.  
Navigate to: http://<phoneIPAddress>/coreConf.htm#lo  
Specify “sip” event logging settings.  
Local  
Navigate to: http://<phoneIPAddress>/appConf.htm#lo  
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.  
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4.5.1 IP TOS  
The “type of service” field in an IP packet header consists of four TOS bits and a 3-bit  
precedence field. Each TOS bit can be set to either 0 or 1. The precedence field can  
be set to a value from 0 through 7. The type of service can be configured specifically  
for RTP packets and call control packets, such as SIP signaling packets.  
Configurationfile: Specify protocol-specific IP TOS settings.  
Central  
ipmid.cfg  
(boot  
•
For more information, see 4.6.1.7.2 IP TOS <IP/> on  
page 87.  
server)  
Web Server (if  
enabled)  
Specify IP TOS settings.  
Navigate to: http://<phoneIPAddress>/coreConf.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  
when it has a valid VLAN ID set in its network configuration, or is instructed to tag  
packets via Cisco Discovery Protocol (CDP) running on a connected Ethernet switch,  
or a VLAN ID is obtained from DHCP (see 2.2.1.2.2 DHCP Menu on page 7). The  
802.1p/Q user_priority field can be set to a value from 0 to 7. The user_priority can be  
configured specifically for RTP packets and call control packets, such as SIP signaling  
packets, with default settings configurable for all other packets.  
Configuration file: Specify default and protocol-specific 802.1p/Q settings.  
Central  
ipmid.cfg  
(boot  
•
For more information, see 4.6.1.7.1 Ethernet IEEE  
802.1p/Q <Ethernet/> on page 86.  
server)  
Web Server (if  
enabled)  
Specify 802.1p/Q settings.  
Navigate to http://<phoneIPAddress>/coreConf.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>Settings>Network  
Configuration  
•
For more information, see 2.2.1 Basic Network Setup on  
page 4.  
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4.5.3 RTCP Support  
The phone supports RTCP per RFC 1889. For each RTP stream, which, by 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 three sections:  
• Core Configuration - ipmid.cfg  
• Application Configuration - sip.cfg  
• Per-phone Configuration - phone1.cfg  
Notes  
In the following tables, “Null” should be interpreted as the empty string, that is, attributeName=“”  
when the file is viewed in a text editor.  
To enter special characters in a configuration file, enter the appropriate sequence using a text editor.  
See the table below.  
Special Character  
Required Character Sequence in Text Editor  
&
”
&amp;  
&quot;  
&apos;  
&lt;  
’
<
>
&gt;  
4.6.1 Core Configuration - ipmid.cfg  
4.6.1.1 Localization <localization/>  
The phone has a multilingual user interface. It supports both North American and  
international time and date formats. The call progress tones can also be customized.  
For more information, see 4.6.1.1.3 Call Progress Tones <callProgTones> on page 68,  
4.6.1.3.2 Chord Sets <chord_sets/> on page 71, and 4.6.1.5.1.1 Call Progress Patterns  
on page 75.  
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4.6.1.1.1 Multilingual <multilingual/>  
The multilingual feature is based on string dictionary files downloaded from the boot  
server. These files are encoded in standalone XML format. Several western European  
and Asian languages are included with the distribution.  
Permitted  
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.cpt.x  
positive integer  
The call progress tone index to be  
associated with this language. See  
4.6.1.1.3 Call Progress Tones <call-  
ProgTones> on page 68.  
lcl.ml.lang.clock.x.24HourClock  
0,1  
If attribute present, overrides  
lcl.datetime.time.24HourClock;  
If 1, display time in 24-hour clock  
mode rather than a.m./p.m.  
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Permitted  
Values  
Attribute  
Interpretation  
lcl.ml.lang.clock.x.format  
string which  
If attribute present, overrides  
lcl.datetime.date.format;  
D = day of week  
includes ‘D’, ‘d’  
and ‘M’ and two  
optional comma’s  
d = day  
M = month  
Up to two comma’s may be included.  
e.g. D,dM = Thursday, 3 July  
Md,D = July 3, Thursday  
The field may contain 0, 1 or 2  
comma’s which can occur only  
between characters and only one at a  
time i.e. “D,,dM” is illegal.  
lcl.ml.lang.clock.x.longFormat  
0, 1  
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  
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.  
4.6.1.1.1.1 Adding New Languages  
Follow these steps to add new languages to those included with the distribution:  
1. Create a new dictionary file based on an existing one.  
2. Change the strings making sure to encode the XML file in UTF-8 but also  
ensuring the UTF-8 characters chosen are within the Unicode character  
ranges indicated in 3.5.1 Multilingual User Interface on page 46.  
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.  
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4.6.1.1.2 Date and Time <datetime/>  
Permitted  
Optimization  
Attribute  
Values  
Interpretation  
lcl.datetime.time.24HourClock  
0,1  
If 1, display time in 24-hour clock mode rather  
than a.m./p.m.  
lcl.datetime.date.format  
string which Controls format of date string.  
includes  
‘D’, ‘d’ and  
D = day of week  
d = day  
‘M’ and two M = month  
optional  
Up to two comma’s may be included.  
comma’s  
e.g. D,dM = Thursday, 3 July  
Md,D = July 3, Thursday  
The field may contain 0, 1 or 2 comma’s which  
can occur only between characters and only one  
at a time i.e. “D,,dM” is illegal.  
lcl.datetime.date.longFormat  
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.  
4.6.1.1.3 Call Progress Tones <callProgTones>  
Call progress tone overrides can be used to customize the tones for a particular country  
or region. The overrides set offered by default spans all default languages on the  
phone. Tone overrides are based on the ITU-T Recommendation E.180 Supplement 2  
entitled Telephone Network and ISDN - Operation, numbering, routing and mobile  
service - Various tones used in national networks.  
Permitted  
Values  
Attribute  
Interpretation  
lcl.cpt  
positive  
integer OR  
blank  
The index of the default tone overrides to be  
selected by the phone. If blank, default call  
progress tones are used.  
lcl.cpt.menu.x  
string  
String to specify the country or region e.g. Italy.  
Multiple lcl.cpt.menu.x strings are supported,  
the strings are displayed in the Call Progress  
Tones menu. The lcl.cpt.menu.x attributes must  
be sequential (lcl.cpt.menu.1, lcl.cpt.menu.2,  
lcl.cpt.menu.3, ..., lcl.cpt.menu.N) with no gaps.  
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In the following table, x is the index of the region as specified by the x index of the  
lcl.cpt.menu.x attribute above, y is the chord set number and cat is one of cp or misc.  
For more information, see 4.6.1.5.1.1 Call Progress Patterns on page 75.  
Permitted  
Values  
Attribute  
Interpretation  
lcl.cpt.chord.cat.x.y.freq.z  
0-1600  
Frequency for this component in Hertz; up to four  
chord-set components can be specified (z=1, 2, 3,  
4).  
lcl.cpt.chord.cat.x.y.level.z  
-57 to 3  
Level of this component in dBm0.  
lcl.cpt.chord.cat.x.y.onDur  
positive  
integer  
On duration in milliseconds, 0=infinite.  
lcl.cpt.chord.cat.x.y.offDur  
positive  
integer  
Off duration in milliseconds, 0=infinite.  
lcl.cpt.chord.cat.x.y.repeat  
positive  
integer  
Specifies how many times the ON/OFF cadence  
is repeated, 0=infinite.  
4.6.1.2 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, handsfree 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 via network signaling.  
up.oneTouchVoiceMail  
0, 1  
If set to 1, the voicemail summary dis-  
play is bypassed and voicemail 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  
4.6.1.3 Tones <tones/>  
This section describes configuration items for the tone resources available in the  
phone.  
4.6.1.3.1 Dual Tone Multi-Frequency <DTMF/>  
Permitted  
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 via  
RFC 2833 format in its Session  
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.  
4.6.1.3.2 Chord Sets <chord_sets/>  
Chord sets are the building blocks of sound effects that use synthesized rather than  
sampled audio (most call progress and ringer sound effects). A chord-set is a multi-  
frequency note with an optional on/off cadence. A chord-set can contain up to four  
frequency components generated simultaneously, each with its own level.  
There are three blocks of chord sets:  
• callProg: used for call progress sound effect patterns  
• ringer  
• misc (miscellaneous)  
All three blocks use the same chord set specification format.  
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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.  
4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/>  
The following sampled audio WAVE file (.wav) formats are supported:  
• mono 8 kHz G.711 µ-Law  
• G.711 A-Law  
• L16/1600 (16-bit, 16 kHz sampling rate, mono)  
The phone uses built-in wave files for some sound effects. The built-in wave files can  
be replaced with files downloaded from the boot server or from the Internet, however,  
these are stored in volatile memory so the files will need to remain accessible should  
the phone need to be rebooted. Files will be truncated to a maximum size of 300 kilo-  
bytes.  
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  
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The following table defines the default usage of the sampled audio files with the  
phone:  
Sampled Audio File  
Default use within phone (pattern reference)  
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  
4.6.1.5 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.  
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4.6.1.5.1 Patterns <patterns/>  
Patterns use a simple script language that allows different chord sets or wave files to  
be strung together with periods of silence. The script language uses the following  
instructions:  
Instruction  
Meaning  
Example  
sampled (n)  
Play sampled audio  
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 instruc-  
optional and allows  
the chord set ON  
duration to be over-  
ridden to d millisec-  
onds)  
tion 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)  
branch (n)  
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)  
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).  
Attribute  
Permitted Values  
Interpretation  
se.pat.callProg.x.name  
UTF-8 encoded  
string  
Used for identification purposes in the user inter-  
face (currently used for ringer patterns only); for  
patterns that use a sampled audio file which has  
been overridden by a downloaded replacement, the  
se.pat.ringer.x.name parameter will be overridden  
in the user interface by the file names of the wave  
file.  
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Attribute  
Permitted Values  
Interpretation  
se.pat.callProg.x.inst.y.type  
sampled OR chord As above.  
OR silence OR  
branch  
se.pat.callProg.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.callProg.x.inst.y.param  
positive integer  
If instruction type is chord, this optional parameter  
specifies the on duration to be used, overriding the  
on duration specified in the chord-set definition.  
4.6.1.5.1.1 Call Progress Patterns  
The following table maps call progress patterns to their usage within the phone.  
Call progress  
pattern number  
Use within phone  
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  
message waiting tone  
alerting  
intercom announcement tone  
barge-in tone  
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4.6.1.5.1.2 Ringer Patterns  
The following table maps ringer pattern numbers to their default descriptions.  
Ringer pattern  
number  
Default description  
a
1
Silent Ring  
2
Low Trill  
3
Low Double Trill  
Medium Trill  
Medium Double Trill  
High Trill  
4
5
6
7
High Double Trill  
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  
Sampled audio file 10  
Sampled audio file 11  
14  
15  
16  
17  
18  
19  
20  
21  
22  
a. Silent Ring will only provide a visual indication of an incoming call, but no audio  
indication.  
b. Sampled audio files 1-21 all use the same built-in file unless that file has been  
replaced with a downloaded file. For more information, see 4.6.1.4 Sampled  
Audio for Sound Effects <sampled_audio/> on page 72.  
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4.6.1.5.1.3 Miscellaneous Patterns  
The following table maps miscellaneous patterns to their usage within the phone.  
Miscellaneous  
pattern number  
Use within phone  
1
2
3
4
5
6
7
new message waiting indication  
new instant message  
not used  
local hold notification  
positive confirmation  
negative confirmation  
welcome (boot up)  
4.6.1.5.2 Ring type <ringType/>  
Ring type is used to define a simple class of ring to be applied based on some 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 .  
b
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.  
b. See note a.  
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.  
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Attribute  
Permitted Values  
Interpretation  
se.rt.modification.enabled 0,1  
Set to 1 to allow user modification via local  
user interface of the pre-defined ring type  
a
enabled for modification .  
se.rt.x.name  
se.rt.x.type  
UTF-8 encoded string  
Used for identification purposes in the user  
b
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  
‘ring’ or ‘ring-answer’  
The ringer index to be used for this class of  
ring. The ringer index should match one of  
4.6.1.5.1.2 Ringer Patterns on page 76.  
se.rt.x.callWait  
integer - only relevant  
if the type is set to  
‘ring’ or ‘ring-answer’  
The call waiting index to be used for this  
class of ring. The call waiting index should  
match one defined in 4.6.1.5.1.1 Call  
Progress Patterns on page 75.  
se.rt.x.timeout  
se.rt.x.mod  
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.  
0,1  
Set to 1 if the user interface should allow for  
modification by the user of the ringer index  
used for this ring class.  
a. Modification via user interface will be implemented in a future release.  
b. Modification via user interface will be implemented in a future release.  
4.6.1.6 Voice Settings <voice/>  
4.6.1.6.1 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  
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4.6.1.6.1.1 Codec Preferences <preferences/>  
Permitted  
Optimization  
Attribute  
Values  
Default Interpretation  
voice.codecPref.G711Mu  
voice.codecPref.G711A  
voice.codecPref.G729AB  
Null, 1-3  
1
2
3
Specifies the codec preferences for  
®
SoundPoint IP platforms.  
1=highest  
3=lowest  
Null=do not use  
Give each codec a unique priority,  
this will dictate the order used in  
SDP negotiations.  
Note: If less than two codecs are  
specified, the phone will automati-  
cally use G711Mu and G711A with  
default parameters.  
voice.codecPref.IP_4000.G711Mu  
voice.codecPref.IP_4000.G711A  
Null, 1-3  
1
2
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.  
4.6.1.6.1.2 Codec Profiles <profiles/>  
The following profile attributes can be adjusted for each of the three supported codecs.  
In the table, x=G711Mu, G711A, or G729AB.  
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.  
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Permitted  
Values  
Attribute  
Interpretation  
voice.audioProfile.x.jitterBufferMin  
40, 50, 60, ... The smallest jitter buffer depth (in milli-  
(multiple of  
10)  
seconds) that must be achieved before  
play out begins for the first time. Once  
this depth has been achieved initially, the  
depth may fall below this point and play  
out will still continue. This parameter  
should be set to the smallest possible  
value which is at least two packet pay-  
loads, and larger than the expected short  
term average jitter.  
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).  
voice.audiProfile.x.jitterBufferMax  
> jitterBuf-  
ferMin,  
multiple of  
10,  
<=500 for IP  
500 and IP  
600,  
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.  
<= 160 for IP  
300  
4.6.1.6.2 Volume Persistence <volume/>  
The user’s selection of the receive volume during a call can be remembered between  
calls. This can be configured per termination (handset, headset and handsfree/chassis).  
In some countries regulations exist which dictate that receive volume should be reset  
to nominal at the start of each call on handset and headset.  
Permitted  
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.  
If set to 0, the receive volume will be  
reset to nominal at the start of each  
call.  
0, 1  
0, 1  
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4.6.1.6.3 Gains <gains/>  
The default gain settings have been carefully adjusted to comply with the TIA-810-A  
digital telephony standard.  
Do not alter these values.  
Attribute  
Default  
0
voice.gain.rx.analog.handset  
voice.gain.rx.analog.headset  
voice.gain.rx.analog.chassis  
0
3
voice.gain.rx.analog.chassis.IP300  
voice.gain.rx.analog.chassis.IP_4000  
voice.gain.rx.analog.ringer  
-6  
3
3
voice.gain.rx.analog.ringer.IP300  
voice.gain.rx.analog.ringer.IP_4000  
voice.gain.rx.digital.handset  
voice.gain.rx.digital.headset  
-6  
3
-15  
-21  
0
voice.gain.rx.digital.chassis  
voice.gain.rx.digital.chassis.IP_4000  
voice.gain.rx.digital.ringer  
0
-21  
-21  
-24  
-24  
3
voice.gain.rx.digital.ringer.IP_4000  
voice.gain.rx.analog.handset.sidetone  
voice.gain.rx.analog.headset.sidetone  
voice.gain.tx.analog.handset  
voice.gain.tx.analog.headset  
voice.gain.tx.analog.chassis  
3
3
voice.gain.tx.analog.chassis.IP_4000  
voice.gain.tx.digital.handset  
3
0
voice.gain.tx.digital.headset  
0
voice.gain.tx.digital.chassis  
6
voice.gain.tx.digital.chassis.IP_4000  
voice.gain.tx.analog.preamp.handset  
voice.gain.tx.analog.preamp.headset  
voice.gain.tx.analog.preamp.chassis  
0
23  
23  
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4.6.1.6.4 Acoustic Echo Cancellation <AEC/>  
These settings control the performance of the speakerphone acoustic echo canceller.  
Do not alter these values.  
Attribute  
Default  
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  
125  
3800  
-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  
125  
3800  
-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  
125  
3800  
-6  
0
0
-6  
-6  
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4.6.1.6.5 Acoustic Echo Suppression <AES/>  
These settings control the performance of the speakerphone acoustic echo suppressor.  
Do not alter these values.  
Attribute  
Default  
voice.aes.hs.enable  
0
7
0
0
1
7
7
6
6
6
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  
4.6.1.6.6 Background Noise Suppression <NS/>  
These settings control the performance of the transmit background noise suppression  
feature.  
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Do not alter these values.  
Attribute  
Default  
voice.ns.hs.enable  
0
voice.ns.hs.signalAttn  
voice.ns.hs.silenceAttn  
voice.ns.hd.enable  
-6  
-8  
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  
-8  
1
-6  
voice.ns.hf.IP_4000.silenceAttn -8  
4.6.1.6.7 Automatic Gain Control <AGC/>  
7
These settings control the performance of the transmit automatic gain control feature.  
Do not alter 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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4.6.1.6.8 Receive Equalization <RXEQ/>  
These settings control the performance of the handsfree receive equalizaton feature.  
Do not alter these values.  
Attribute  
Default  
voice.rxEq.hf.IP_500.enable  
voice.rxEq.hf.IP_600.enable  
voice.rxEq.hf.IP_4000.enable  
voice.rxEq.hf.type  
1
1
0
0
voice.rxEq.hf.IP_4000.type  
voice.rxEq.hf.size  
0
63  
voice.rxEq.hf.IP_4000.size  
voice.rxEq.hf.coef.0 to .62  
voice.rxEq.hf.coef.IP_4000.0 to .32  
33  
Don’t change these values  
Don’t change these values  
4.6.1.6.9 Transmit Equalization <TXEQ/>  
These settings control the performance of the handsfree transmit equalization feature.  
Do not alter these values.  
Attribute  
Default  
voice.txEq.hf.IP_500.enable  
voice.txEq.hf.IP_600.enable  
voice.txEq.hf.IP_4000.enable  
voice.txEq.hf.coef.1 to 20  
voice.txEq.hf.coef.IP_4000.1 to 20  
0
0
0
Don’t change these values  
Don’t change these values  
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4.6.1.6.10 Voice Activity Detection <VAD/>  
These settings control the performance of the voice activity detection (silence 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.  
4.6.1.7 Quality of Service <QOS/>  
These settings control the Quality of Service (QOS) options.  
4.6.1.7.1 Ethernet IEEE 802.1p/Q <Ethernet/>  
These settings control the 802.1p/Q user_priority field.  
4.6.1.7.1.1 RTP <RTP/>  
These parameters apply to RTP packets.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ethernet.rtp.user_priority 0-7  
5
User-priority used for RTP packets.  
4.6.1.7.1.2 Call Control <CallControl/>  
These parameters apply to call control packets, such as the network protocol signaling.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ethernet.callControl.user_priority  
0-7  
5
User-priority used for call con-  
trol packets.  
86  
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4.6.1.7.1.3 Other <Other/>  
These default parameter values are used for all packets which are not set explicitly.  
Permitted  
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.  
4.6.1.7.2 IP TOS <IP/>  
These settings control the “type of service” field in outgoing packets.  
4.6.1.7.2.1 RTP <RTP/>  
These parameters apply to RTP packets.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
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.  
qos.ip.rtp.max_throughput  
qos.ip.rtp.max_reliability  
qos.ip.rtp.min_cost  
0, 1  
0, 1  
0, 1  
0-7  
1
0
0
5
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.  
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.  
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4.6.1.7.2.2 Call Control <CallControl/>  
These parameters apply to call control packets, such as the network protocol signaling.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
qos.ip.callControl.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.  
qos.ip.callControl.max_throughput  
qos.ip.callControl.max_reliability  
qos.ip.callControl.min_cost  
0, 1  
0, 1  
0, 1  
0-7  
0
0
0
5
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.  
qos.ip.callControl.precedence  
If set to 1, set precedence bits in  
the IP TOS field of the IP  
header, or else don’t set them.  
4.6.1.8 Basic TCP/IP <TCP_IP/>  
4.6.1.8.1 Network Monitoring <netMon/>  
Do not alter these values.  
Attribute  
Permitted Values  
0, 1  
Default  
tcpIpApp.netMon.enabled  
tcpIpApp.netMon.period  
1
1 to 86400  
30  
4.6.1.8.2 Time Synchronization <SNTP/>  
The following table describes the parameters used to set up time synchronization and  
daylight savings time. The defaults shown will enable daylight savings time for  
North America.  
Daylight savings defaults:  
• don’t use fixed day, use first or last day of week in the month,  
• start DST on the first Sunday in April at 2 am,  
88  
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• 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 SNTP re-  
syncs.  
a
valid host  
name or IP  
address  
clock  
Address of the SNTP  
server.  
tcpIpApp.sntp.address  
tcpIpApp.sntp.gmtOffset  
positive or  
negative  
integer  
-28800  
(Pacific  
time)  
Offset in seconds of the  
local time zone from  
GMT.  
Note: 3600 seconds per  
hour  
tcpIpApp.sntp.daylightSavings.enable  
0, 1  
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, “April 1st” is  
used, otherwise “the  
first Sunday in April” is  
used.  
tcpIpApp.sntp.daylightSavings.start.month  
1-12  
4 (April)  
Month to start DST.  
1=Jan, 2=Feb, ...,  
12=Dec  
tcpIpApp.sntp.daylightSavings.start.date  
tcpIpApp.sntp.daylightSavings.start.time  
1-31  
0-23  
1
2
Day of the month to  
start DST.  
Time of day to start  
DST, in 24 hour clock.  
2=2 am, 14=2 pm  
tcpIpApp.sntp.daylightSavings.start.dayOf- 1-7  
Week  
1
0
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  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
tcpIpApp.sntp.daylightSavings.stop.date  
1-31  
1
Day of the month to  
start DST.  
tcpIpApp.sntp.daylightSavings.stop.time  
0-23  
2
1
1
Time of day to stop  
DST, in 24 hour clock.  
2= 2 am, 14=2 pm  
tcpIpApp.sntp.daylightSavings.stop.dayOf-  
Week  
1-7  
Day of week to stop  
DST. 1=Sun, 2=Mon,  
..., 7=Sat  
tcpIpApp.sntp.daylightSavings.stop.dayOf-  
Week.lastInMonth  
0, 1  
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.  
a. Both tcpIpApp.sntp.address and tcpIpApp.sntp.gmtOffset can be provided via DHCP. If so,  
the DHCP parameters will override the parameters in ipmid.cfg.  
4.6.1.8.3 port <port/>  
4.6.1.8.3.1 RTP <RTP/>  
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 (via 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 (via 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 tcpIpApp.port.rtp.filter-  
ByIp and tcpI-  
pApp.port.rtp.filterByPort  
must be enabled for this to  
work.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
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.  
4.6.1.9 Web Server <HTTPD/>  
The phone contains a local web server for user and administrator features. This can be  
disabled for applications where it is not needed or where it poses a security threat. The  
web server supports both basic and digest authentication. The authentication user  
name and password are not configurable for this release.  
Permitted  
Values  
Attribute  
Default  
Interpretation  
httpd.enabled  
0, 1  
1
If set to 1, the HTTP server will be enabled.  
4.6.1.9.1 Configuration <cfg/>  
Permitted  
Values  
Attribute  
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.  
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4.6.1.10 Call Handling Configuration <call/>  
Permitted  
Optimization  
Attribute  
Values  
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  
call.offeringTimeOut  
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.  
positive  
integer  
60  
Time in seconds to allow an incoming call  
to ring before dropping the call, 0=infi-  
a
nite .  
call.ringBackTimeOut  
call.lastCallReturnString  
call.callWaiting.prompt  
positive  
integer  
60  
*69  
0
Time in seconds to allow an outgoing call  
to remain in the ringback state before  
dropping the call, 0=infinite.  
string of  
maximum  
length 32  
The string sent to the server when the user  
selects the “last call return” action.  
0, 1  
If set to 1, an incoming call received when  
another call is active will change the User  
Interface focus (call appearance and soft  
keys).  
call.shared.disableDivert  
0, 1  
0, 1  
0, 1  
1
1
0
If set to 1, disable diversion feature for  
shared lines.  
call.shared.seizeFailReorder  
If set to 1, play re-order tone locally on  
shared line seize failure.  
call.shared.one-  
TouchResume  
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 automatically.  
If set to 0, pressing the shared line will  
bring up a list of the calls on that line and  
the user can select which call the next  
action should be applied to.  
a. The call diversion, no answer feature will take precedence over this feature if enabled. For  
more information, see 4.6.3.3.3 No Answer <noanswer/> on page 125.  
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4.6.1.10.1 Hold, Local Reminder <localReminder/>  
Permitted  
Optimization  
Attribute  
Values  
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.  
4.6.1.11 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.  
8. The phone could have 2 megabytes or 4 megabytes of flash memory depending on the hardware model.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
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.  
Maximum size in Kbytes of  
non-volatile storage that the  
directory will be permitted to  
consume.  
1 to 100  
100  
Maximum size in Kbytes of  
volatile storage that the direc-  
tory will be permitted to con-  
sume.  
4.6.1.12 Fonts <font/>  
This section does not apply to the IP 300.  
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, see 3.5.1 Multilingual  
User Interface on page 46.  
• <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 mangled 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 500  
“fontProp_10”  
This is the font used widely in the current implementation.  
This is the soft key specific font.  
“fontPropSoftkey_10”  
SoundPoint® IP 600  
“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 mangled .fnt blocks.  
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4.6.1.12.1 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.names are defined, the  
index x must follow consecutive  
increasing order.  
4.6.1.12.2 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.names are defined, the  
index x must follow consecutive  
increasing order.  
4.6.1.13 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, 500 and 600 key functions can be changed from the factory  
defaults, although this is typically not necessary. For each key whose function you  
wish to change, add an XML attribute in the format described in the following table to  
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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_500 or IP_600, y is the key number. IP 300: 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  
Usable in application  
ArrowDown  
ArrowLeft  
Yes  
Yes  
Yes  
Yes  
No  
Yes  
No  
No  
No  
No  
No  
No  
Yes  
No  
No  
No  
No  
No  
ArrowRight  
ArrowUp  
Browse  
BuddyStatus  
CallAppearance1  
CallAppearance2  
CallAppearance3  
CallAppearance4  
CallAppearance5  
CallAppearance6  
CallList  
CallPark  
CallPickup  
CallWaiting  
Callers  
Cancel  
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Function  
ColdBoot  
Conference  
Delete  
Usable in application  
No  
Yes  
Yes  
No  
Dial  
Dialpad0  
Dialpad1  
Dialpad2  
Dialpad3  
Dialpad4  
Dialpad5  
Dialpad6  
Dialpad7  
Dialpad8  
Dialpad9  
DialpadStar  
DialpadPound  
Directories  
Divert  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
No  
DoNotDisturb  
Email  
Yes  
No  
Favorites  
Flash  
No  
No  
Handsfree  
Headset  
Feature  
Yes  
Yes  
No  
Help  
No  
Hold  
Yes  
Yes  
Yes  
Yes  
Yes  
Line1  
Line2  
Line3  
Line4  
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Function  
Line5  
Usable in application  
Yes  
Yes  
No  
Line6  
LocalDirectory  
Macro  
No  
Messages  
Menu  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
No  
MicMute  
MyStatus  
Null  
Offline  
Redial  
Release  
RemoteDirectory  
Select  
No  
Yes  
No  
SelfTest  
Services  
Setup  
No  
Yes  
No  
Shift  
SoftKey1  
SoftKey2  
SoftKey3  
SoftKey4  
SpeedDial  
SpeedDialMenu  
Suspend  
Transfer  
VoiceMail  
VolDown  
VolUp  
Yes  
Yes  
Yes  
Yes  
Yes  
Yes  
No  
Yes  
No  
Yes  
Yes  
No  
WarmBoot  
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4.6.1.14 Bitmaps <bitmaps/>  
Bitmaps used by the phone are defined in this section.  
4.6.1.14.1 Platform <IP_300/>, <IP_500/>, <IP_600/> and  
<IP_4000/>  
In the following table, x=IP_300, IP_500, IP_600, or IP_4000, y is the bitmap number.  
Attribute  
Permitted Values  
Interpretation  
bitmap.x.y.name  
The bitmap name  
representing a bit-  
map to be used.  
This is a bitmap name to be used for assignment when  
doing animation. If the bitmap should be downloaded  
from the boot server, the bitmap name must be:  
1. Different from any name already in usage in  
ipmid.cfg.  
2. Must match the name of the corresponding <file-  
Name>.bmp expected to be retrieved from the  
boot server.  
4.6.1.15 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  
ipmid.cfg.  
100  
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4.6.1.15.1 Animations <Animations/> <IP_300/>, <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_500, IP_600 or IP_4000, y is the animation number, z is  
the step in the animation.  
Attribute  
Permitted Values  
Interpretation  
ind.anim.x.y.frame.z.bitmap  
A bitmap name  
defined previously.  
Bitmap to use. Note that it must be  
defined already, see 4.6.1.14.1 Platform  
<IP_300/>, <IP_500/>, <IP_600/> and  
<IP_4000/> on page 100  
ind.anim.x.y.frame.z.duration  
positive integer  
Duration in milliseconds for this step.  
0=infinite.  
4.6.1.15.2 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.  
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4.6.1.15.3 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 pattern  
index, i.e. index x in the <Patterns/> tag above. For  
GraphicIcon type indicators, index refers to anima-  
tion index, i.e. index y in the <Animations/> tag  
above.  
4.6.1.15.4 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.  
4.6.1.15.4.1 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 above) for this indicator.  
This maps logical index to a specific physical LED.  
ind.led.x.physNum  
4.6.1.15.4.2 Graphic Icons <gi/> <IP_300/>, <IP_500/>, <IP_600/> and  
<IP_4000/>  
In the following table, x=IP_300, IP_500, IP_600 or IP_4000, y is the graphic icon  
number.  
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).  
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Permitted  
Values  
Attribute  
Interpretation  
ind.gi.x.y.class  
ind.gi.x.y.physX  
positive integer Assigns the class (defined above) for this indicator.  
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 500: 0-159  
IP 600: 0-319  
IP 4000: 0-247  
ind.gi.x.y.physY  
ind.gi.x.y.physW  
ind.gi.x.y.physH  
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 500: 0-79  
IP 600: 0-159  
IP 4000: 0-67  
IP 300: n/a  
For GraphicIcon type indicators, this is the width of the  
indicator measured in pixels.  
IP 500: 1-160  
IP 600: 1-320  
IP 4000: 1-248  
IP 300: n/a  
For GraphicIcon type indicators, this is the height of the  
indicator measured in pixels.  
IP 500: 1-80  
IP 600: 1-160  
IP 4000: 1-68  
4.6.1.16 Event Logging <logging/>  
Logging parameter changes can impair system operation. Do not change any logging parameters  
without prior consultation with Polycom.  
Warning!  
The event logging system supports the following classes of events:  
Level  
Interpretation  
0
1
2
3
Debug only  
High detail event class  
Moderate detail event class  
Low detail event class  
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Level  
Interpretation  
4
5
6
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  
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  
4.6.1.16.1 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.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
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
See above table for timestamp  
type.  
log.render.realtime  
log.render.stdout  
log.render.file  
0, 1  
0, 1  
0, 1  
1
Set to 1. Do not change.  
Set to 1. Do not change.  
Set to 1. Do not change.  
1
1
log.render.file.size  
positive  
integer  
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.  
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.  
4.6.1.16.2 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  
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log task is controlled by a unique attribute set starting with log.sched.x where x identi-  
fies 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  
4.6.1.17 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) via protocol signaling.  
4.6.1.17.1 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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4.6.1.18 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  
Note: Changing these param-  
eters is not advised.  
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.  
4.6.1.19 RAM Disk <RAMdisk/>  
These settings control the phone’s internal RAM disk feature. Changing these param-  
eters 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.  
4.6.1.20 Request <request/>  
4.6.1.20.1 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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4.6.1.21 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)  
•
•
•
•
•
•
•
•
•
“cpt-settings”,  
“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)  
“cpt-settings” controls whether call progress  
tones can be selected by the phone user using  
the Settings menu  
•
•
•
“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)  
If set to 0, the feature will be disabled.  
If set to 1, the feature will be enabled and usable by  
the local user.  
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4.6.1.22 Resource <resource/>  
These settings control the maximum size or an external resource retrieved at run time.  
4.6.1.22.1 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.  
4.6.1.22.2 quotas <quotas/>  
Permitted  
Values  
Attribute  
Interpretation  
res.quotas.x.name  
“tone”, “bit-  
The name of the sub-application for which the particu-  
map”, “font”, or lar quota will apply:  
“xmlui”  
•
“tone” relates to all downloaded tones and sound  
effects  
•
•
•
“bitmap” relates to all downloaded bitmaps  
“font” relates to all downloaded fonts  
“xmlui” relates to XML driven user interface  
a
available on some platforms  
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.  
a. This is available on MGCP application, but not on SIP application.  
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4.6.1.23 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>  
4.6.1.23.1 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.  
4.6.1.23.2 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  
Null or 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. 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  
Null or 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.  
Increasing this value may have a detri-  
mental effect on performance of the  
phone.  
112  
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4.6.2 SIP Configuration - sip.cfg  
The configuration file sip.cfg contains SIP protocol settings that would typically apply  
to an entire installation and must be set before the phones will be operational, unless  
changed via the local web server interface or local menu settings on the phone. Set-  
tings include the local port used for SIP signaling, the address and ports of a cluster of  
SIP servers, and other parameters. The following sections describe each of these  
parameters.  
4.6.2.1 Protocol <volpProt/>  
4.6.2.1.1 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.  
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4.6.2.1.2 Server <server/>  
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.  
voIpProt.server.x.transport  
DNSnaptr or  
DNSna If set to Null or DNSnaptr:  
TCPpreferred or ptr  
UDPonly  
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.  
voIpProt.server.x.expires  
positive integer,  
minimum 300  
3600  
Requested registration period  
a
in seconds .  
114  
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Permitted  
Values  
Attribute  
Default Interpretation  
voIpProt.server.x.register  
0, 1  
1
If set to 0, calls can be routed to  
an outbound proxy without reg-  
istration.  
voIpProt.server.x.retryTimeOut  
Null or  
non-negative  
integer  
0
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).  
voIpProt.server.x.retryMaxCount  
voIpProt.server.x.expires.lineSeize  
Null or  
non-negative  
integer  
3
If set to 0 or Null, 3 is used.  
retryMaxCount retries will be  
attempted before moving on to  
the next available server.  
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 when half the negotiated period has  
expired.  
4.6.2.1.3 SIP <SIP/>  
Permitted  
Values  
Attribute  
Default Interpretation  
voIpProt.SIP.useRFC2543hold  
0, 1  
1
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, 1  
0
0
If set to 1, the proprietary “epid”  
parameter is added to the From field of  
all requests to support Windows Live  
Communications Server.  
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’.  
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Permitted  
Values  
Attribute  
Default Interpretation  
®
voIpProt.SIP.WM50  
0, 1  
0
If set to 1, Windows Messenger 5.0  
will be supported.  
®
If set to 0, Windows Messenger 4.7  
will be supported.  
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 endpoint 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:.  
4.6.2.1.3.1 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  
4.6.2.1.3.2 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 (see 4.6.1.5.2  
Ring type <ringType/> on  
page 77) will be applied.  
requests  
voIpProt.SIP.alertInfo.x.class  
positive  
integer  
Null  
116  
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4.6.2.1.3.3 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  
“INVITE”,  
“ACK”, “BYE”,  
“REGISTER”,  
“CANCEL”,  
“OPTIONS”,  
“INFO”,  
a
applied .  
“MESSAGE”,  
“SUB-  
SCRIBE”,  
“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.  
a. WARNING: Intensive request validation may have a negative performance impact due to  
the additional signaling required in some cases, therefore, use it judiciously.  
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4.6.2.1.3.4 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.  
4.6.2.1.3.5 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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4.6.2.1.4 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.  
4.6.2.1.4.1 Digit Map <digitmap/>  
Attribute  
Permitted Values  
Default  
Interpretation  
dialplan.digitmap  
string compatible with [2-9]11|0T|  
When this attribute is  
present, number-only  
dialing during the setup  
phase of new calls will  
the digit map feature  
of MGCP described in  
2.1.5 of RFC 3435.  
011xxx.T|  
[0-1][2-  
9]xxxxxxxxx|  
String is limited to 512 [2-9]xxxxxxxxx| be compared against the  
bytes and 20 seg-  
[2-9]xxxT  
patterns therein and if a  
match is found, the call  
will be initiated automat-  
ically eliminating the  
need to press Send.  
ments; a comma is  
also allowed; when  
reached in the digit  
map, a comma will  
turn dial tone back on.  
dialplan.digitmap.timeOut positive integer  
3
Timeout in seconds for  
‘T’ feature of digitmap.  
4.6.2.1.4.2 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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4.6.2.1.4.2.1 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. Multi-  
ple servers can be listed start-  
ing with x=1, 2, ... for fault  
tolerance.  
dialplan.rout-  
ing.server.x.port  
1 to 65535  
5060  
4.6.2.1.4.2.2 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  
4.6.2.1.4.2.1 Server  
<server/> on page 120  
that will be used for  
emergency routing.  
4.6.2.1.5 SIP Logging <logging/> <level/> <change/>  
Permitted  
Values  
Attribute  
Default Interpretation  
log.level.change.sip  
0-5  
4
This attribute controls the logging detail level  
for the SIP component. These are the input fil-  
ters for the internal memory-based log system.  
120  
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4.6.3 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 config-  
uration files. For more information, see 2.2.2.1.3 Deploying a Boot Server for the  
Phones on page 13.  
4.6.3.1 Registration <reg/>  
SoundPoint® IP 300 supports two unique registrations, IP 500 supports three and IP  
600 supports six. 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: x=1-2; IP 500: x=1-3; IP  
600: x=1-6; IP 4000: x=1  
Permitted  
Values  
Attribute  
Default Interpretation  
reg.x.displayName  
UTF-8 encoded  
string  
Null  
Display name used for local user  
interface as well as SIP signaling.  
reg.x.address  
string in the for- Null  
mat userPart or  
user-  
The actual address corresponding  
to this registration (userPart or  
[email protected]). For user  
part only registration  
Part@domain  
(reg.x.address=”1002”), the regis-  
tration will be userPart@proxy-  
HostNameOrIPAddress where  
userPart is reg.x.address and proxy-  
HostNameOrIPAddress is either  
reg.x.server.1.address if non-Null  
or voIpProt.server.1.address speci-  
fied in sip.cfg.  
reg.x.label  
reg.x.type  
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.  
private OR  
shared  
private If set to private, use standard call  
signaling.  
If set to shared, augment call sig-  
naling with call state subscriptions  
and notifications and use access  
control for outgoing calls.  
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Permitted  
Values  
Attribute  
Default Interpretation  
reg.x.thirdPartyName  
string in the  
same format as  
reg.x.address  
Null  
This field must match the  
reg.x.address value of the other  
registration which makes up the  
bridged line.  
reg.x.auth.userId  
string  
string  
Null  
User ID to be used for authentica-  
tion challenges for this registration.  
If non-Null, will override the “Reg  
User x” parameter entered into the  
Authentication submenu off of the  
Settings menu on the phone.  
reg.x.auth.password  
reg.x.server.y.address  
Null  
Null  
Password to be used for authentica-  
tion challenges for this registration.  
If non-Null, will override the “Reg  
Password x” parameter entered into  
the Authentication submenu off of  
the Settings menu on the phone.  
dotted-decimal  
IP address or  
host name  
Optional IP address or host name,  
port, transport, registration period,  
fail-over parameters and lineseize  
subscription period of a SIP server  
that accepts registrations. Multiple  
servers can be listed starting with  
y=1, 2, ... for fault tolerance. If  
specified, these servers will over-  
ride the servers specified in sip.cfg  
in 4.6.2.1.2 Server <server/> on  
page 114.  
reg.x.server.y.port  
0, Null, 1 to  
65535  
Null  
Null  
reg.x.server.y.transport  
DNSnaptr or  
TCPpreferred or  
UDPonly  
reg.x.server.y.expires  
positive integer  
0, 1  
Null  
Null  
NOTE:  
reg.x.server.y.register  
reg.x.server.y.retryTimeOut  
If the reg.x.server.y.address param-  
eter is non-Null, all of the  
reg.x.server.y.xxx parameters will  
override the parameters specified in  
sip.cfg in 4.6.2.1.2 Server <server/  
> on page 114  
Null or non-neg- Null  
ative integer  
reg.x.server.y.retryMaxCount  
Null or non-neg- Null  
ative integer  
reg.x.server.y.expires.lineSeize positive integer  
Null  
reg.x.acd-login-logout  
0, 1  
0, 1  
0
0
If both parameters are set to 1 for a  
registration, the ACD feature will  
be enabled for that registration.  
reg.x.acd-agent-available  
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4.6.3.2 Calls <call/>  
These sections describe call-oriented per-phone configuration items.  
4.6.3.2.1 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.  
4.6.3.2.2 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: x=1-2; IP 500: x=1-3; IP  
600: x=1-6; 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])  
4.6.3.2.3 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.  
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In the following table, x is the registration number. IP 300: x=1-2; IP 500: x=1-3; IP  
600: x=1-6; 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.  
4.6.3.3 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: x=1-2; IP 500: x=1-3; IP  
600: x=1-6; 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 (see 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.  
124  
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4.6.3.3.1 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 via the soft key menu.  
4.6.3.3.2 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.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]  
4.6.3.3.3 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  
divert.noanswer.x.contact  
positive integer  
60  
Time in seconds to allow  
altering before initiating the  
diversion.  
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.  
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4.6.3.3.4 Do Not Disturb <dnd/>  
The phone can automatically divert calls when 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.  
4.6.3.4 Dial Plan <dialplan/>  
Per-registration dial plan configuration is supported. In the following tables, x is the  
registration number. IP 300: x=1-2; IP 500: x=1-3; IP 600: x=1-6; 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, see  
4.6.2.1.3.5 Conference Setup  
<conference/> on page 118.  
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, see  
4.6.2.1.3.5 Conference Setup  
<conference/> on page 118.  
126  
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4.6.3.4.1 Digit Map <digitmap/>  
Permitted  
Optimization  
Attribute  
Values  
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  
digit map, a comma will  
turn dial tone back on.  
For more information, see  
4.6.2.1.3.5 Conference  
Setup <conference/> on  
page 118.  
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, see  
4.6.2.1.3.5 Conference  
Setup <conference/> on  
page 118.  
4.6.3.4.2 Routing <routing/>  
This configuration section allows specific routing paths for outgoing SIP calls to be  
configured independent of other ‘default’ configuration.  
4.6.3.4.2.1 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  
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4.6.3.4.2.2 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  
4.6.3.4.2.2 Emergency  
<emergency/> on  
page 128 that will be  
used for emergency rout-  
ing.  
4.6.3.5 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. See  
3.6.1 Voicemail Integration on page 49.  
Instant Messages will still be accessible  
from the Main Menu.  
128  
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4.6.3.5.1 Message Waiting Indicator <mwi/>  
In the following table, x is the registration number. IP 300: x=1-2; IP 500: x=1-3; IP  
600: x=1-6; 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  
“registration”  
for x = 1,  
“disabled” for  
others  
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)  
4.6.3.6 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.  
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Permitted  
Values  
Attribute  
Default  
Interpretation  
nat.signalPort  
1024 to  
65536  
Null  
If non-Null, this port will be used by the phone  
for SIP signaling, overriding the value set for  
voIpProt.local.signalPort in sip.cfg.  
nat.mediaPortStart  
1024 to  
65536  
Null  
If non-Null, this attribute will be used to set the  
initially allocated RTP port, overriding the  
value set for tcpIpApp.port.rtp.mediaPortRang-  
eStart in ipmid.cfg. See 4.6.1.8.3.1 RTP <RTP/  
> on page 90.  
130  
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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 (see  
list below for supported RFC’s and drafts). Transfer is included in the basic SIP sup-  
port.  
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  
draft-ietf-sip-replaces-03.txt  
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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Session Initiation Protocol (SIP)  
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  
132  
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Session Initiation Protocol (SIP)  
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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Session Initiation Protocol (SIP)  
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.  
134  
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Session Initiation Protocol (SIP)  
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  
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  
420 Bad Extension  
No  
No  
Yes  
No  
No  
421 Extension Required  
423 Interval Too Brief  
No  
No  
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Session Initiation Protocol (SIP)  
Response  
Supported  
Yes  
Yes  
Yes  
No  
Notes  
480 Temporarily Unavailable  
481 Call/Transaction Does Not Exist  
482 Loop Detected  
483 Too Many Hops  
484 Address Incomplete  
485 Ambiguous  
Yes  
No  
486 Busy Here  
Yes  
No  
487 Request Terminated  
488 Not Acceptable Here  
491 Request Pending  
493 Undecipherable  
No  
No  
No  
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  
5.1.4.6 6xx Responses - Global Failure  
Response  
Supported  
No  
Notes  
600 Busy Everywhere  
603 Decline  
Yes  
604 Does Not Exist Anywhere No  
136  
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Session Initiation Protocol (SIP)  
Response  
Supported  
Notes  
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 direc-  
tions with the “a” SDP media attributes sendonly, recvonly, inactive or sendrecv. The  
hold signaling method used by the phone is configurable (for more information, see  
4.6.2.1.3 SIP <SIP/> on page 115) but both methods are supported when signaled by  
the remote endpoint.  
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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Session Initiation Protocol (SIP)  
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)  
draft-levy-sip-diversion-04.txt  
draft-ietf-sip-session-timer-12.txt  
draft-ietf-sipping-mwi-02.txt  
Diversion Indication in SIP  
Session Timers in the Session Initiation Protocol (SIP)  
A Message Summary and Message Waiting Indication  
Event Package for the Session Initiation Protocol (SIP)  
draft-ietf-sipping-dialog-package-03.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  
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.  
138  
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Administrator Guide - SoundPoint® IP / SoundStation® IP  
Session Initiation Protocol (SIP)  
5.2.3 SIP for Instant Messaging and Presence Leverag-  
ing Extensions (SIMPLE)  
The phone is compatible with the Presence and Instant Messaging features of Win-  
dows® Messenger® and MSN® Messenger 4.7 and Windows® Messenger® 5.0. In a  
future release, support for the Presence and Instant Message recommendations in the  
SIP 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  
endpoints 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  
5.2.5 Bridged Line Appearance Signaling  
A bridged line is an address of record managed by a server. The server allows multi-  
ple endpoints 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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